Re: [asterisk-users] How to have failover sip interface?
On Sun, Oct 24, 2010 at 12:02 PM, sean darcy seandar...@gmail.com wrote: My asterisk machine has 2 nic's. One nic (ETH0) is connected to a cable modem. The other nic (ETH1) is connected to an internal lan. The internal lan also has access to the internet. The cable service, Time-Warner RoadRunner, is great when up, but is not reliable. And sip connections are excellent. The connection through the internal lan (Verizon DSL) is reliable but lousy. Sigh. When the cable is down, the interface connection to the cable modem stays up. An ifconfig shows ETH0 as up. The only way to tell is to ping an outside address. I thought of bonding. But that won't work since it will see ETH0 as up, even if the cable service is down. Is there a way to implement network failover that actually checks for true internet connection? This way I can keep my sip connection up, even if degraded. sean -- _ -- 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 To be honest, I'm not sure Asterisk could do that, but I could be wrong. I've tried to fool with multiple interfaces in the past and as I recall Asterisk can only bond to 1 interface. However, if anyone else knows different I'd LOVE to find that they changed that :D -- _ -- 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] asterisk 1.6 and Firefox 4 Beta
Has anyone started using Firefox 4 beta versions? We started today and find that many of the GUI's attached to Asterisk respond differently and in many cases not at all? We have found that details cannot be saves and that the screens become very unstable. While we appreciate this is a beta Firefox it would appear they have deviated from their 3.x format with regards to interfacing. Rupert Utteridge -- _ -- 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] BLF in Asterisk 1.4.*
Hello everybody, does anybody know if BLF is correctly working in Asterisk 1.4.*? I'm particularly interested in Asterisk 1.4.25. Thanks in advance! Phil -- _ -- 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] trixbox - sip trunk with voipwise
Hi, No matter I try, I can not register to Voipwise with Trixbox. It is always in unregistered state in sip registry. Here is my last sip trunk configuration: PEER DETAILS: allow=g729 bindport=5060 disallow=alldtmfmode=rfc2833 fromdomain=sip.voipwise.com fromuser=username host=sip.voipwise.com insecure=very maxexpirey=120 pickupgroup=1 port=5060 secret=pass type=peer username=username Register string: username:p...@sip.voipwise.com username%3ap...@sip.voipwise.com Do you have any suggestions? Thank you. Mert -- _ -- 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] Asterisk 1.6 Overlap dialling timeout?
Hello, I'm experimenting with Overlap Dialling in asterisk 1.6. I've enabled this in sip.conf and on the SNOM 300 phone. My problem is that asterisk dials out as soon as it matches an extension without waiting to see if the user is going to type in more digits. Is there a way to set a timeout per channel or globally? I'd like Asterisk to wait for a few seconds once its found a match in case the user needs to key in more digits. Thank You. Regards, Veselin K -- _ -- 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] BLF in Asterisk 1.4.*
Yes, it works fine in 1.4.22 and 1.4.27 and 1.4.35. Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) On 2010-10-29 5:15 AM, Asterisk User an.asterisk.u...@gmail.com wrote: Hello everybody, does anybody know if BLF is correctly working in Asterisk 1.4.*? I'm particularly interested in Asterisk 1.4.25. Thanks in advance! Phil -- _ -- 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
Re: [asterisk-users] SIP client floods port 5060 and gets blocked
Hello, any more input on this subject ?! Kind regards, Jonas. Original Message Subject:Re: [asterisk-users] SIP client floods port 5060 and gets blocked Date: Thu, 28 Oct 2010 13:42:12 +0200 From: Jonas Kellens jonas.kell...@telenet.be To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com On 10/28/2010 12:52 PM, Gordon Henderson wrote: On Thu, 28 Oct 2010, Jonas Kellens wrote On 10/28/2010 10:44 AM, Kevin Keane wrote: I assume that you checked and the remote IP is a legitimate IP phone? If not, it could be an attempt to break into your system. If it is a legitimate IP phone, make sure that the SIP configuration is correct -- if the SIP authentication fails, you can see this happening. 1. This is a legitimate phone, yes. 2. Registration goes as follow : REGISTER SIP/2.0 401 Unauthorized Re-Register with Digest 200 OK Is it s Snom phone? I've seen Snoms do this... Gordon I have this with Snom 320, Snom 370, Grandstream GXW4008 and YeaLink T28... Jonas. -- _ -- 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] Re : thousands Hangup per second /saturation of bandwidth
On Fri, Oct 29, 2010 at 6:47 AM, Per Jessen p...@computer.org wrote: Sherwood McGowan wrote: [snip] I'm still at a loss as to why people a) get pissy because no-one responds to a help request on an all volunteer list, Often it is because they are not aware it is an all-volunteer list. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. -- _ -- 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 I guess the name doesn't tip anyone off -- _ -- 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] MGCP
Hi! I have asterisk 1.4 I want to make a MGCP trunk as a client to connect to a provider who is using MGCP protocol, he provided me with user password, You will most probably need 1.8 for this, with 1.4 you will certainly not be able to succed. Read more: http://www.voip-info.org/wiki/view/Asterisk+MGCP+channels Quote Matthew Fredrickson: A new channel driver, called chan_ccs, that allows, among other things, you to control MGCP media gateways for bearer trunks, instead of having to locally terminate them on the asterisk box that's controlling the signaling links. There is also code in the same branch that has chan_ccs that modified chan_mgcp so that Asterisk can act as a media gateway (since I don't have any good real media gateways to test on). This basically means you can have Asterisk TDM channel scalability up to (in the ideal state) the same level as you can do with SIP with no media, per box. Philipp -- _ -- 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] New tutorial: Compiling Asterisk 1.8 on CentOS 64
Hello all, as everybody else here - I guess - I have been playing with the new Asterisk 1.8 release. So far everything went smoothly - the compilation phase was really straightforward, and I have a box ready for real testing now. I prepared a tutorial out of my experience on how to compile Asterisk 1.8 with iCal, GTalk, SNMP, MySQL, cURL and DAHDI - the usual stuff - so if anybody is interested or has suggestions/improvements, it's here: http://astrecipes.net/index.php?n=398 .I did not include H323 this time as I don't have H.323 gear anymore to test it with! :) Comments are welcome. l. -- Loway - home of QueueMetrics - http://queuemetrics.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
[asterisk-users] Asterisk 1.8 and character sets and AMI
Hi, Just tried upgrading to 1.8 and ran into two problem immediately; 1. Caller-ID behavior is different -- now when I set the caller-id name to something with special characters (Ö, for example), the SIP INVITE now has %C3%96 instead of the Ö character. I've tried doing Set(CALLERID(name-charset)=utf8) as well as iso8859-1, but it's always the same behavior. 2. My AMI scripts have stopped working and Asterisk console shows a Broken Pipe error. Has the I/O to AMI changed? I had a quick glance through the change log and couldn't find anything indicating different. Haven't started looking at what the output looks like, but it would be nice if someone could point me to a document going through the changes so I don't have to re-invent the wheel. Anyone have any info on either one? Best regards, Örn Arnarson -- _ -- 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] Video based Asterisk Training
Hi Friends, We have created a video based training for Asterisk in English and Urdu. Please check them and let us know how we can improve them for no-voice users. http://www.youtube.com/watch?v=KXq9g8UiGnQ http://www.youtube.com/watch?v=MID2RvgdD7s http://www.youtube.com/watch?v=_LbDUdAGfSY http://www.youtube.com/watch?v=J9Chkrg7E-M http://www.youtube.com/watch?v=MsC12wc9ZnU http://www.youtube.com/watch?v=Y12exIN1soY More You can find here http://www.youtube.com/supertecacademy Thank You -- Regards, Zuhair Raza Asst. Manager Technical Support Phone: +1-850-433-8555 ext 101064 Website: www.didx.net Skype: zuhairraza What is DIDX.net? http://www.youtube.com/watch?v=mIgGTGkTZns http://www.youtube.com/watch?v=mIgGTGkTZns -- _ -- 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] Video based Asterisk Training
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zuhair Raza Sent: Friday, October 29, 2010 4:14 PM To: asterisk-users@lists.digium.com; asterisk-...@lists.digium.com Subject: [asterisk-users] Video based Asterisk Training Hi Friends, We have created a video based training for Asterisk in English and Urdu. Please check them and let us know how we can improve them for no-voice users. http://www.youtube.com/watch?v=KXq9g8UiGnQ http://www.youtube.com/watch?v=KXq9g8UiGnQ http://www.youtube.com/watch?v=MID2RvgdD7s http://www.youtube.com/watch?v=MID2RvgdD7s http://www.youtube.com/watch?v=_LbDUdAGfSY http://www.youtube.com/watch?v=_LbDUdAGfSY http://www.youtube.com/watch?v=J9Chkrg7E-M http://www.youtube.com/watch?v=J9Chkrg7E-M http://www.youtube.com/watch?v=MsC12wc9ZnU http://www.youtube.com/watch?v=MsC12wc9ZnU http://www.youtube.com/watch?v=Y12exIN1soY http://www.youtube.com/watch?v=Y12exIN1soY More You can find here http://www.youtube.com/supertecacademy Thank You -- Regards, Zuhair Raza Asst. Manager Technical Support Phone: +1-850-433-8555 ext 101064 Website: www.didx.net Skype: zuhairraza I watched the sip.conf video and have this suggestion - you need to put the text you are talking about on the screen. The point of watching a youtube video is to get audio and visual feedback. If I wanted just the audio, I'd listen to a podcast. The format of your video now is sip.conf Iax.conf Extensions.conf (video) Blah-blah-blah (audio) Make it like this sip.conf Iax.conf Extensions.conf (video - opening) (video for reference) Sip.conf snippet Extensions.conf snippet Iax.conf snippet Same audio. -- _ -- 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] Updating asteriskcdrdb with uniqueid field from Master.csv, Master.csv.1....Master.csv.5 - Must I bring all files together first?
Hi Everyone, Just noted that PBXinaFLASH failed me again on yet something else. The uniqueid field didn't update on the asteriskcdrdb in the past few months but it is available in the .csv files in /var/log/asterisk/cdr-csv/*.csv I have already re-did the asterisk-addons install with correct parameters to include future calls uniqueid in the table (I have no clue why these flavours of Asterisk chose to remove neccessary parameters from time to time) Anyhow, the uniqueid field is needed to be there for ARI to work and for recordings to be pulled. So, I am checking the folder for .csv files and I see Master.csv.1 Master.csv.2 Master.csv.3..Master.csv.5. My Questions: 1- Must I bring those files into one big file first before trying to update the MySQL table? 2- Must I remove all data from asteriskcdrdb first to avoid duplication and then do the update through a php script? Thanks -- _ -- 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] Video based Asterisk Training
Thanks Danny for your comments, they are covered later in next videos. I elaborated config files with sections and settings in the next videos and also the ending video is in process will upload it tomorrow. On Sat, Oct 30, 2010 at 2:44 AM, Danny Nicholas da...@debsinc.com wrote: -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Zuhair Raza *Sent:* Friday, October 29, 2010 4:14 PM *To:* asterisk-users@lists.digium.com; asterisk-...@lists.digium.com *Subject:* [asterisk-users] Video based Asterisk Training Hi Friends, We have created a video based training for Asterisk in English and Urdu. Please check them and let us know how we can improve them for no-voice users. http://www.youtube.com/watch?v=KXq9g8UiGnQ http://www.youtube.com/watch?v=MID2RvgdD7s http://www.youtube.com/watch?v=_LbDUdAGfSY http://www.youtube.com/watch?v=J9Chkrg7E-M http://www.youtube.com/watch?v=MsC12wc9ZnU http://www.youtube.com/watch?v=Y12exIN1soY More You can find here http://www.youtube.com/supertecacademy Thank You -- Regards, Zuhair Raza Asst. Manager Technical Support Phone: +1-850-433-8555 ext 101064 Website: www.didx.net Skype: zuhairraza I watched the sip.conf video and have this suggestion – you need to put the text you are talking about on the screen. The point of watching a youtube video is to get audio and visual feedback. If I wanted just the audio, I’d listen to a podcast. The format of your video now is “sip.conf Iax.conf Extensions.conf” (video) Blah-blah-blah (audio) Make it like this “sip.conf Iax.conf Extensions.conf” (video – opening) (video for reference) Sip.conf snippet Extensions.conf snippet Iax.conf snippet Same audio. -- _ -- 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 -- Regards, Zuhair Raza Asst. Manager Technical Support Phone: +1-850-433-8555 ext 101064 Website: www.didx.net Skype: zuhairraza What is DIDX.net? http://www.youtube.com/watch?v=mIgGTGkTZns http://www.youtube.com/watch?v=mIgGTGkTZns -- _ -- 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] Video based Asterisk Training
On Fri, Oct 29, 2010 at 5:02 PM, Zuhair Raza z...@supertec.com wrote: Thanks Danny for your comments, they are covered later in next videos. I elaborated config files with sections and settings in the next videos and also the ending video is in process will upload it tomorrow. On Sat, Oct 30, 2010 at 2:44 AM, Danny Nicholas da...@debsinc.com wrote: From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zuhair Raza Sent: Friday, October 29, 2010 4:14 PM To: asterisk-users@lists.digium.com; asterisk-...@lists.digium.com Subject: [asterisk-users] Video based Asterisk Training Hi Friends, We have created a video based training for Asterisk in English and Urdu. Please check them and let us know how we can improve them for no-voice users. http://www.youtube.com/watch?v=KXq9g8UiGnQ http://www.youtube.com/watch?v=MID2RvgdD7s http://www.youtube.com/watch?v=_LbDUdAGfSY http://www.youtube.com/watch?v=J9Chkrg7E-M http://www.youtube.com/watch?v=MsC12wc9ZnU http://www.youtube.com/watch?v=Y12exIN1soY More You can find here http://www.youtube.com/supertecacademy Thank You -- Regards, Zuhair Raza Asst. Manager Technical Support Phone: +1-850-433-8555 ext 101064 Website: www.didx.net Skype: zuhairraza I watched the sip.conf video and have this suggestion – you need to put the text you are talking about on the screen. The point of watching a youtube video is to get audio and visual feedback. If I wanted just the audio, I’d listen to a podcast. The format of your video now is “sip.conf Iax.conf Extensions.conf” (video) Blah-blah-blah (audio) Make it like this “sip.conf Iax.conf Extensions.conf” (video – opening) (video for reference) Sip.conf snippet Extensions.conf snippet Iax.conf snippet Same audio. -- _ -- 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 -- Regards, Zuhair Raza Asst. Manager Technical Support Phone: +1-850-433-8555 ext 101064 Website: www.didx.net Skype: zuhairraza What is DIDX.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 I love the idea, primarily because I'm in the middle of producing a series of training videos as well. I look forward to seeing the finished product :) -- _ -- 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
