[asterisk-users] Trevlig lampa :)
http://xn--stilbyrn-g0a.se/salt/industrilampa-brottbyhttp://stilbyrån.se/salt/industrilampa-brottby -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: esk...@gmail.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] Skall vi gå på Smaklösa?
http://www.facebook.com/smaklosa?sk=events //Mattias -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: esk...@gmail.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
Re: [asterisk-users] Not Dialing 9
Way make it complicated. Make shore Asterisk match internal numbers first. Else external number. Trixbox works like that by default. No nead for 9 to call external numbers. //Mattias On Fri, Jan 9, 2009 at 12:07 AM, Thczv F. Thczv thczv.th...@gmail.comwrote: When I set up my Asterisk box at home I didn't want to have to dial 9 to dial off premises, so I gave all my local phones three digit extensions with this format: 1[1,0]*. My thought is that there are no area codes that start with 0 or 1, so if I use those numbers, I can create 20 local extensions that can be dialed with 3 digits, and not have to use a timeout when dialing long distance. If I dial 1, then anything other than 0 or 1, Asterisk knows I am dialing long distance. If I start with any number other than 1, Asterisk knows I am dialing a local or local toll call. This has worked fine for me (as far as I know). Is there some flaw I am not seeing? I see a lot of small businesses that require a 9 to dial out, even though they don't have very many extensions. Couldn't they do what I did and not have to dial 9? I ask because we are having a problem where I work with our Cisco 7940 phones adding an extra 1 sometimes, which gets the local Sheriff upset (too many 911 calls). Thanks, Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: esk...@gmail.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP to IAX?
Hi all! I am looking for some software that would work as a proxy between a SIP device (SIP phones and ATA boxes) and the Asterisk system running IAX. The reason is that I can only get IAX trow the firewalls, and can't change the settings. One solution I am using are getting several Asterisk system communicate trow IAX bout in this case would I rater have every persons computer act as a proxy for their own phones (Running Widows XP). The reason is that the are using laptops and travel, some are already using softphons and IAX bout some don't like softphons for some reason. If it is not any proxy out their, the will I write o of my own. (Of cause giving it out for free), I think Asterisk for Windows would be overkill. Sorry for my poor English. Regards Mattias Andersson Sweden ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Max amount of concurrent calls on and iax trunk
I agree bandwidth is the limit, however the reason to use IAX is it is saving bandwidth. I am runging 2 Trixbox CE with IAX over a 2 Mbit line. I have never had any isues with the IAX trunk. I wish that I could get son good IAX phones to the office, tan would we skip on Trixbox and run the phones directly over a VPN Chanel. SIP over VPN are giving more hassle then IAX sound wise. On Thu, Aug 7, 2008 at 1:45 AM, Chris Brentano [EMAIL PROTECTED] wrote: I have two Asterisk 1.4 boxes connected via IAX over a VPN tunnel on a 10Mbit link. We never did any stress testing as it's a temporary arrangement, but we've never had any call quality issues or run up against concurrent ca, holl limitations. I'm mostly routing internal extensions over the trunk, and in the case of two floating users I have their extensions at each office ring when their DID is called. One server is an older Pentium 4 1.7 GHz with 1GB Ram, and the other is a Dual Xeon 2.33 GHz with 4GB Ram. As for codec, I'm disallowing all except ulaw and gsm, with ulaw the priority codec for hardphones (Polycom) and gsm the priority for softphones (X-Lite, Zoiper). I would expect the limitation you're going to run up against is not Asterisk, but the bandwidth between your two systems. On 6 Aug, 2008, at 10:40 AM, Rosli Sukri wrote: hi, wanted to ask if anybody has experienced setting up two asterisk 1.2 boxes connected via iax trunk. have u guys ever stress tested the trunks i.e how many concurrent calls can a trunk handle and whether codec has any effect on it. ATT1.c ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Communication between two asterisk server
I am using IAX2, easier to get to work trow firewalls. //Mattias On Fri, Feb 15, 2008 at 1:14 PM, [EMAIL PROTECTED] wrote: Hi Bhrugu , Thanks for the reply. I will check it off. Regards, Preeta -Original Message- From: [EMAIL PROTECTED] on behalf of Bhrugu Mehta Sent: Fri 2/15/2008 5:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Communication between two asterisk server hi,preeta you have to change sip.conf in both server. suppose, server 1 and server 2 both are asterisk server. you want to call from server 1 to server 2. then, in ser-1, sip.conf [general] register= user:[EMAIL PROTECTED] [user] type=friend fromuser=user username=user secret=pass host=ipofserver2 context=any in server2, sip.conf [user] type=friend username=user secret=user host=dynamic context=anyyouwant Bhrugu Mehta (SAI INFO SYSTEM LTD.) On 2/15/08, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi, I want that an sjphone registered using serverA can call to an sjphone registered using serverB and vice vers. I want to know how two asterisk server communicate to each other. Please let me know, for that, what configuration file I have to change. Thanking you, Regards, Preeta Pandey The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Lithuania
Hi All! Maybe a little of topic. Bout coming from Sweden and needing to call Lithuania a lot am I wondering if anyone on the list could recommend a sheep service in Lithuania to connect my Asterisk to. A local number are not necessary bout preferd for incoming calls for my contacts. Regards Mattias Andersson Adress: Mattias Andersson Storskiftesvägen 6 S-145 60 Norsborg Mobil: +46-70-799 44 41 Email: [EMAIL PROTECTED] Skype: eskes1 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Block some number outgoing from joust oneextention
Hi! Unfortunately did this stop Asterisk to register ny phones and trunk. Did I put tit in the wrong place? //Mattias Hi! Exactly what I needed. It was the 209 part that I did not figure put. Thanks! //Mattias At 03:53 2007-01-05, you wrote: exten = _9070X./209,1,NoOP,SORRY CHARLIE exten = _9070X./209,2,Congestion This would block any call from 209 to 070X as long as 9 was your outside digit. I use the NoOP to help me out with the CLI and debugging :) Hope this helps Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mattias Andersson Sent: Thursday, January 04, 2007 5:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Block some number outgoing from joust oneextention Thanks! I can´t rely figure out how to block for only one extension. Eg. Extension 209 need to be blocked from making calls starting with 070 (eg. 9070). Some clues did I get bout would it men a new form-internal-blocked dialplan? Regards Mattias On 04/01/07, C F mailto:[EMAIL PROTECTED][EMAIL PROTECTED] wrote: The easiest way is thru using contexts. On 1/3/07, Mattias Andersson mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi all! I am shore someone have writing about it bout I cant find it. I have a extension that I need to block from making expansive mobil calls. Everyone else should be aloud to do the calls. I am shore it is possible to be done sens I had a commercial asterisk based PBX that I did that on. However I have switch to Trixbox because I need some custom functions not supported by the commercial product. I would appreciate all help. Regards Mattias Adress: Mattias Andersson Storskiftesvägen 6 S-145 60 Norsborg Mobil: +46-70-799 44 41 Email: mailto:[EMAIL PROTECTED][EMAIL PROTECTED] Skype: eskes1 ___ --Bandwidth and Colocation provided by http://Easynews.comEasynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by http://Easynews.comEasynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: mailto:[EMAIL PROTECTED][EMAIL PROTECTED] -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.432 / Virus Database: 268.16.4/615 - Release Date: 1/3/2007 1:34 PM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Adress: Mattias Andersson Storskiftesvägen 6 S-145 60 Norsborg Mobil: +46-70-799 44 41 Email: [EMAIL PROTECTED] Skype: eskes1 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Block some number outgoing from joust oneextention
Hi! Exactly what I needed. It was the 209 part that I did not figure put. Thanks! //Mattias At 03:53 2007-01-05, you wrote: exten = _9070X./209,1,NoOP,SORRY CHARLIE exten = _9070X./209,2,Congestion This would block any call from 209 to 070X as long as 9 was your outside digit. I use the NoOP to help me out with the CLI and debugging :) Hope this helps Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mattias Andersson Sent: Thursday, January 04, 2007 5:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Block some number outgoing from joust oneextention Thanks! I can´t rely figure out how to block for only one extension. Eg. Extension 209 need to be blocked from making calls starting with 070 (eg. 9070). Some clues did I get bout would it men a new form-internal-blocked dialplan? Regards Mattias On 04/01/07, C F mailto:[EMAIL PROTECTED][EMAIL PROTECTED] wrote: The easiest way is thru using contexts. On 1/3/07, Mattias Andersson mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi all! I am shore someone have writing about it bout I cant find it. I have a extension that I need to block from making expansive mobil calls. Everyone else should be aloud to do the calls. I am shore it is possible to be done sens I had a commercial asterisk based PBX that I did that on. However I have switch to Trixbox because I need some custom functions not supported by the commercial product. I would appreciate all help. Regards Mattias Adress: Mattias Andersson Storskiftesvägen 6 S-145 60 Norsborg Mobil: +46-70-799 44 41 Email: mailto:[EMAIL PROTECTED][EMAIL PROTECTED] Skype: eskes1 ___ --Bandwidth and Colocation provided by http://Easynews.comEasynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by http://Easynews.comEasynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: mailto:[EMAIL PROTECTED][EMAIL PROTECTED] -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.432 / Virus Database: 268.16.4/615 - Release Date: 1/3/2007 1:34 PM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Adress: Mattias Andersson Storskiftesvägen 6 S-145 60 Norsborg Mobil: +46-70-799 44 41 Email: [EMAIL PROTECTED] Skype: eskes1 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Block some number outgoing from joust one extention
Thanks! I can´t rely figure out how to block for only one extension. Eg. Extension 209 need to be blocked from making calls starting with 070 (eg. 9070). Some clues did I get bout would it men a new form-internal-blocked dialplan? Regards Mattias On 04/01/07, C F [EMAIL PROTECTED] wrote: The easiest way is thru using contexts. On 1/3/07, Mattias Andersson [EMAIL PROTECTED] wrote: Hi all! I am shore someone have writing about it bout I cant find it. I have a extension that I need to block from making expansive mobil calls. Everyone else should be aloud to do the calls. I am shore it is possible to be done sens I had a commercial asterisk based PBX that I did that on. However I have switch to Trixbox because I need some custom functions not supported by the commercial product. I would appreciate all help. Regards Mattias Adress: Mattias Andersson Storskiftesvägen 6 S-145 60 Norsborg Mobil: +46-70-799 44 41 Email: [EMAIL PROTECTED] Skype: eskes1 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Block some number outgoing from joust one extention
Hi all! I am shore someone have writing about it bout I cant find it. I have a extension that I need to block from making expansive mobil calls. Everyone else should be aloud to do the calls. I am shore it is possible to be done sens I had a commercial asterisk based PBX that I did that on. However I have switch to Trixbox because I need some custom functions not supported by the commercial product. I would appreciate all help. Regards Mattias Adress: Mattias Andersson Storskiftesvägen 6 S-145 60 Norsborg Mobil: +46-70-799 44 41 Email: [EMAIL PROTECTED] Skype: eskes1 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ISA server Issue (Maybe off topic)
Hi! I have my Trixbox running behind a ISA server. However it works fine with Rix telecom (The service provider) The same setup dos not allow my phone trow the ISA server. It is seeing the phone as registering the public adress of the firewall instead of port forwarding it. anyone else had this issue? I am suing Sjphone and X-lite //Mattias Adress: Mattias Andersson Storskiftesvägen 6 S-145 60 Norsborg Mobil: +46-70-799 44 41 Email: [EMAIL PROTECTED] Skype: eskes1 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitoring an asterisk server during off hours
Somewhere did I see a test script. I will see if I can find it once more. With that information should you be able to write a simple script that monitor the server and then will notify you if the server stop responding. PING wold maybe also be a help. //Mattias On 29/11/06, Olivier [EMAIL PROTECTED] wrote: Hi, During off hours, a server of mine simply forward incoming calls to an outside number, so that no user is locally available to report or notify downtimes. As availability is here a major requirement, I'm looking for a cost effective and reliable way to monitor this server. Should I simply call every 10 minutes, a dedicated extension to check PSTN lines and server availability or is there a smarter way to do it ? Setup: Nagios - Monitoring Asterisk -- PSTN -- Monitored Asterisk - VPN access --www --- Back to Monitoring Asterisk With this a single check would test PSTN lines, asterisk server and VPN access availability. I don't think it should be very difficult to trigger a call from Nagios. Regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7970 SIP upgrade issues
Hi! I have only used 7940 and 7905. The 7940 are supporting TFTP and I did use that to upgrade them. I had to do it in 3 steps. First a old SIP firmware. Then an newer firmware and then the on that I am using. //Mattias On 29/11/06, Paul [EMAIL PROTECTED] wrote: Does anyone have any ideas? I am pulling my hair out :-) I changed email address's which is why the names different. Thanks in advance - Original Message - *From:* Admin @ TheAdmiralNelson.Com [EMAIL PROTECTED] *To:* asterisk-users@lists.digium.com *Sent:* Thursday, November 23, 2006 6:54 PM *Subject:* [asterisk-users] Cisco 7970 SIP upgrade issues Dear Asterisk People, I am having problems putting a SIP image on a 7970. I was wondering if anyone can help? First problem is the phone is running version Load IDJar70.2-5-47-17.sbn Boot Load ID7970_64054100.bin Version5.0(0.6S) So I did read that you couldn't simply put the latest SIP image on such an old phone and a newer firmware version should be used. I got cmterm-7970_7971-sccp.7-0-2SR1 However I can't figure out how to update the firmware without a Callmanager. Can anyone enlighten me? If I do that I can then put the latest SIP image on I think Best Regards -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7970 SIP upgrade issues
Hi Paul! I do thing you could use a TFTP bout I have not ben woring with that phone. Could you post your TFTP loog? //Mattias On 29/11/06, Paul A Brown [EMAIL PROTECTED] wrote: Hi Mattias, That is what I did for my 7960 and what I need to do for this. However my problem is when I un tar the cisco file it won't run. I think it needs call manager :-( Paul - Original Message - *From:* Mattias Andersson [EMAIL PROTECTED] *To:* Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com *Sent:* Wednesday, November 29, 2006 11:26 AM *Subject:* Re: [asterisk-users] Cisco 7970 SIP upgrade issues Hi! I have only used 7940 and 7905. The 7940 are supporting TFTP and I did use that to upgrade them. I had to do it in 3 steps. First a old SIP firmware. Then an newer firmware and then the on that I am using. //Mattias On 29/11/06, Paul [EMAIL PROTECTED] wrote: Does anyone have any ideas? I am pulling my hair out :-) I changed email address's which is why the names different. Thanks in advance - Original Message - *From:* Admin @ TheAdmiralNelson.Com [EMAIL PROTECTED] *To:* asterisk-users@lists.digium.com *Sent:* Thursday, November 23, 2006 6:54 PM *Subject:* [asterisk-users] Cisco 7970 SIP upgrade issues Dear Asterisk People, I am having problems putting a SIP image on a 7970. I was wondering if anyone can help? First problem is the phone is running version Load IDJar70.2-5-47-17.sbn Boot Load ID7970_64054100.bin Version5.0(0.6S) So I did read that you couldn't simply put the latest SIP image on such an old phone and a newer firmware version should be used. I got cmterm-7970_7971-sccp.7-0-2SR1 However I can't figure out how to update the firmware without a Callmanager. Can anyone enlighten me? If I do that I can then put the latest SIP image on I think Best Regards -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: [EMAIL PROTECTED] -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7970 SIP upgrade issues
Hi believe that you nead a standalone image. Would you consider use SIP image, that could be possible to find on the net. //Mattias On 29/11/06, Paul A Brown [EMAIL PROTECTED] wrote: Hi Its not even at the tftp stage. When I run the image file from Chisco and attempt to run setup I get a registry error. I am assuming its because its expecting a call manager. How do I upgrade the firmware? The image I have is only for callmanager cmterm-7970_7971-sccp.7-0-2SR1 Anyone know of a standalone image? - Original Message - *From:* Mattias Andersson [EMAIL PROTECTED] *To:* Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com *Sent:* Wednesday, November 29, 2006 12:41 PM *Subject:* Re: [asterisk-users] Cisco 7970 SIP upgrade issues Hi Paul! I do thing you could use a TFTP bout I have not ben woring with that phone. Could you post your TFTP loog? //Mattias On 29/11/06, Paul A Brown [EMAIL PROTECTED] wrote: Hi Mattias, That is what I did for my 7960 and what I need to do for this. However my problem is when I un tar the cisco file it won't run. I think it needs call manager :-( Paul - Original Message - *From:* Mattias Andersson [EMAIL PROTECTED] *To:* Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com *Sent:* Wednesday, November 29, 2006 11:26 AM *Subject:* Re: [asterisk-users] Cisco 7970 SIP upgrade issues Hi! I have only used 7940 and 7905. The 7940 are supporting TFTP and I did use that to upgrade them. I had to do it in 3 steps. First a old SIP firmware. Then an newer firmware and then the on that I am using. //Mattias On 29/11/06, Paul [EMAIL PROTECTED] wrote: Does anyone have any ideas? I am pulling my hair out :-) I changed email address's which is why the names different. Thanks in advance - Original Message - *From:* Admin @ TheAdmiralNelson.Com [EMAIL PROTECTED] *To:* asterisk-users@lists.digium.com *Sent:* Thursday, November 23, 2006 6:54 PM *Subject:* [asterisk-users] Cisco 7970 SIP upgrade issues Dear Asterisk People, I am having problems putting a SIP image on a 7970. I was wondering if anyone can help? First problem is the phone is running version Load IDJar70.2-5-47-17.sbn Boot Load ID7970_64054100.bin Version5.0(0.6S) So I did read that you couldn't simply put the latest SIP image on such an old phone and a newer firmware version should be used. I got cmterm-7970_7971-sccp.7-0-2SR1 However I can't figure out how to update the firmware without a Callmanager. Can anyone enlighten me? If I do that I can then put the latest SIP image on I think Best Regards -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: [EMAIL PROTECTED] -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: [EMAIL PROTECTED] -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Feature Codes won't work
Hi all! I get problem with *11, *12 for instance. The won't work. I get a message that the phone extension can't be fund for *11 and for *12 will I get A Error. Any idea? //Mattias -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Backup and mail on trixox
Hi all! Is it anyone that have set up a solution where the schedule backup are sent by mail ore FTP automatic? //Mattias -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Caller ID in Sweden not working and looking for and voices
Hi!The problem is that I have commercial Asterisk baste switch that it works wit.My trixbox do not. I guess it has to do with the setting of system for Caller ID.//Mattias On 15/11/06, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote: Am Mittwoch, den 15.11.2006, 01:06 +0100 schrieb Mattias Andersson: Hi! I am getting inbound caller ID fine bout not out. I am in Sweden and suing Rixtelcom /POrt80 as provider. anyone knowing what is wrong? Assuming that is a SIP provider, it is not your job to set the calleridbut the provider's - their interfacing to the regular landline networkis responsible. There are providers that never send callerid, some send always (united, gmx in Germany) and some allow the user to set in hispreferences wether he wants to send callerid (sipgate.de).BRAnselm___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Mattias AnderssonStorskiftesvägen 6145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Caller ID in Sweden not working and looking for and voices
Hi!The problem is that I have commercial Asterisk baste switch that it works wit.My trixbox do not. I guess it has to do with the setting of system for Caller ID.//Mattias On 15/11/06, Mattias Andersson [EMAIL PROTECTED] wrote: Hi!The problem is that I have commercial Asterisk baste switch that it works wit.My trixbox do not. I guess it has to do with the setting of system for Caller ID.//Mattias On 15/11/06, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote: Am Mittwoch, den 15.11.2006, 01:06 +0100 schrieb Mattias Andersson: Hi! I am getting inbound caller ID fine bout not out. I am in Sweden and suing Rixtelcom /POrt80 as provider. anyone knowing what is wrong? Assuming that is a SIP provider, it is not your job to set the calleridbut the provider's - their interfacing to the regular landline networkis responsible. There are providers that never send callerid, some send always (united, gmx in Germany) and some allow the user to set in hispreferences wether he wants to send callerid ( sipgate.de).BRAnselm___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Mattias Andersson Storskiftesvägen 6145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: [EMAIL PROTECTED] -- Mattias AnderssonStorskiftesvägen 6145 60 Norsborg m. +46-70-799 44 41h. +46-8-641 38 97 Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Add Apps to Asterisk?
Hi!The problem is that I have commercial Asterisk baste switch that it works wit.My trixbox do not. I guess it has to do with the setting of system for Caller ID.//Mattias Sorry if I have made double post! (Difficult to verify if mail was sent).On 15/11/06, Darryl Dunkin [EMAIL PROTECTED] wrote:First, check for app_meetme.so in /usr/lib/asterisk/modules (wherever your modules path is). Next, in the CLI, do a 'show modules' to see ifit is there. If not, check your modules.conf and add in 'load =app_meetme.so' assuming autoload is not enabled.-Original Message- From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED] ] On Behalf Of MatthewRubensteinSent: Tuesday, November 14, 2006 21:09To: Asterisk-UsersSubject: [asterisk-users] Add Apps to Asterisk?I've got an Asterisk (v1.2.11) installation running, but it doesn'tseem to have the Meetme() app. At the CLI, I type Meetme , and itresponds No such command 'Meetme'; meetme doesn't show up in CLI showmodules . I'm running a SIP-only server at a datacenter where I can't add Digium (or any other) HW, and am running under CentOS. There isan /etc/asterisk/meetme.conf file, but I don't see anything to use it.What do I have to do, exactly, to install Meetme? What about the Conference command, or others not installed? I'd prefer to use theCentOS package system as much as possible, but I can compile source ifnecessary. Is there a HowTo on the Web somewhere that details thisprocess? --(C) Matthew Rubenstein___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Mattias AnderssonStorskiftesvägen 6145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Caller ID in Sweden not working and looking for and voices
Hi! The problem is that I have commercial Asterisk baste switch that it works wit. My trixbox do not. I guess it has to do with the setting of system for Caller ID. //Mattias I do not now way, but my posting are not coming trow. Or are the? //Mattias On 15/11/06, Mattias Andersson mailto:[EMAIL PROTECTED][EMAIL PROTECTED] wrote: Hi! The problem is that I have commercial Asterisk baste switch that it works wit. My trixbox do not. I guess it has to do with the setting of system for Caller ID. //Mattias On 15/11/06, Anselm Martin Hoffmeister mailto:[EMAIL PROTECTED][EMAIL PROTECTED] wrote: Am Mittwoch, den 15.11.2006, 01:06 +0100 schrieb Mattias Andersson: Hi! I am getting inbound caller ID fine bout not out. I am in Sweden and suing Rixtelcom /POrt80 as provider. anyone knowing what is wrong? Assuming that is a SIP provider, it is not your job to set the callerid but the provider's - their interfacing to the regular landline network is responsible. There are providers that never send callerid, some send always (united, gmx in Germany) and some allow the user to set in his preferences wether he wants to send callerid (http://sipgate.de sipgate.de). BR Anselm ___ --Bandwidth and Colocation provided by http://Easynews.comEasynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: mailto:[EMAIL PROTECTED][EMAIL PROTECTED] -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: mailto:[EMAIL PROTECTED][EMAIL PROTECTED] Adress: Mattias Andersson Storskiftesvägen 6 S-145 60 Norsborg Mobil: +46-70-799 44 41 Email: [EMAIL PROTECTED] Skype: eskes1 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Caller ID in Sweden not working and looking for and voices
Hi!I am getting inbound caller ID fine bout not out.I am in Sweden and suing Rixtelcom /POrt80 as provider.anyone knowing what is wrong?Also is anyone knowing about Swedish voices to trixbox/Asterisk? I have male now and am looking fro female voices. RegardsMattias-- Mattias AnderssonStorskiftesvägen 6145 60 Norsborg m. +46-70-799 44 41h. +46-8-641 38 97 Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Caller ID in Sweden not working and looking for and voices
Hi!I am getting inbound caller ID fine bout not out.I am in Sweden and suing Rixtelcom /POrt80 as provider.anyone knowing what is wrong?Also is anyone knowing about Swedish voices to trixbox/Asterisk? I have male now and am looking fro female voices. Sorry if I have missed a previous answer on the question.RegardsMattias-- Mattias AnderssonStorskiftesvägen 6145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
