Re: [asterisk-users] Cisco IP 8841 asterisk integration
TrueAgree. :) On Mon, Dec 5, 2016 at 11:37 PMwrote: > > True agree, problem is somehow the people purchased am > > supporting to overcome that. Trying level best... around 20 > > phones has been purchased > > Ah, yes, the "we purchased these without consulting you, but it is up to > you to make them work" school of thought. It often goes with, "Well, what > are we paying you for?" and "It's a phone, it shouldn't take you long to > make it work." > > I have to say, unless I am working with a Cisco phone system, Cisco phones > are not my favorite beasts to work with. > __ > This email has been scanned by the Symantec Email Security.cloud service. > For more information please visit http://www.symanteccloud.com > __ > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MOBILE SIMCARD ON ASTERISK
Hi Chris, This would require either an external (or possible internal) GSM/SIM Voip Gateway. Google: "VOIP Gateway GSM Converter SIP IP Phone Adapter" or similar for options. I see some for around $60 US Basically, you'll want to send SIP invite(s), and other commands to and from this device (probably on it's local address, for security, and it acts just as any SIP provider would. Thanks, *Glenn @ VDO* On Mon, Dec 5, 2016 at 6:36 PM, christopher kamutumwa < chriskamutu...@gmail.com> wrote: > Is it possible to have a simcard configured and become incoming line > and outgoing on asterisk and also have the IVR function? If yes wat > hardware is required to have this Accompished > > Thanks > > > Chris > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk. > org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MOBILE SIMCARD ON ASTERISK
Is it possible to have a simcard configured and become incoming line and outgoing on asterisk and also have the IVR function? If yes wat hardware is required to have this Accompished Thanks Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to send dummy audio stream while recording
On Mon, Dec 05, 2016 at 03:29:09PM -0400, Joshua Colp wrote: > On Mon, Dec 5, 2016, at 03:26 PM, jrun wrote: > There is an option, transmit_silence, which can be set in asterisk.conf > that will transmit silence when recording and doing other things. > > -- > Joshua Colp thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to send dummy audio stream while recording
On Mon, Dec 5, 2016, at 03:26 PM, jrun wrote: > > hello, > since while recoding asterisk is not sending an audio stream, the remote > party times-out rtp and is hanging up on us. is it possible to send a > blank > audio stream while recording app is running? There is an option, transmit_silence, which can be set in asterisk.conf that will transmit silence when recording and doing other things. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to send dummy audio stream while recording
hello, since while recoding asterisk is not sending an audio stream, the remote party times-out rtp and is hanging up on us. is it possible to send a blank audio stream while recording app is running? thanks, jrun -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP 8841 asterisk integration
> True agree, problem is somehow the people purchased am > supporting to overcome that. Trying level best... around 20 > phones has been purchased Ah, yes, the "we purchased these without consulting you, but it is up to you to make them work" school of thought. It often goes with, "Well, what are we paying you for?" and "It's a phone, it shouldn't take you long to make it work." I have to say, unless I am working with a Cisco phone system, Cisco phones are not my favorite beasts to work with. __ This email has been scanned by the Symantec Email Security.cloud service. For more information please visit http://www.symanteccloud.com __-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP 8841 asterisk integration
True agree, problem is somehow the people purchased am supporting to overcome that. Trying level best... around 20 phones has been purchased On Mon, 5 Dec 2016, 8:55 p.m. Victor Villarreal,wrote: > With all the money you plan to invest in firmware, licenses, etc., you > have bought a Grandstream IP phone or Yealink... > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP 8841 asterisk integration
With all the money you plan to invest in firmware, licenses, etc., you have bought a Grandstream IP phone or Yealink... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP 8841 asterisk integration
Actually now I have the phones with SIP firmware. I will try with 3pcc firmware along with XML files. Or any idea if we have CUCM application can we change the firmware. am ready to buy the developer edition. Regards . On Mon, 5 Dec 2016, 3:34 p.m. Steve Davies,wrote: > I tried... repeatedly... I failed. I bought some 3PCC phones, and they > just worked. > > If you have the relevant Cisco telephony server product you might be able > to trick it into doing what you want, as that has the proper upgrader for > that model of phone. > > I previously had experience of upgrading the Cisco build to the SIP build > on Cisco 7641 handsets, which have 2 similar builds, but none of the > techniques seemed to apply this time around. > > Cheers, > Steve > > > On Sun, 4 Dec 2016 at 16:03 Gopalakrishnan N > wrote: > > Can't I upload the 3PCC firmware ? available from the Cisco website? > > Actually it came with sip88xx firmware. > > Regards . > > > On Fri, 2 Dec 2016, 10:38 p.m. Steve Davies, wrote: > > Hi, > > You have to buy the 3PCC version for this to work. Once you have this, > they work very much like the Cisco SPA handsets. > > I also ended up with a non-3PCC handset and it is useless, and as far as I > can tell they cannot be re-flashed. > > Cheers, > Steve > > > > On Fri, 2 Dec 2016 at 16:16 Gopalakrishnan N > wrote: > > Anyone tried integrating Cisco IP 8841 phone with Asterisk 11.x. I have > the phone with sip firmware came along with sip88xx-11.0.1SR xx. I tried to > upload woth TFTP due to some reason it's getting failed. Do I need to load > 3pcc firmware or anyway to Configure from the phone itself or from the > GUI? > > I have the SEPMAC.cnf.xml as well. > > Any suggestions would be appreciated. > > Regards . > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP 8841 asterisk integration
I tried... repeatedly... I failed. I bought some 3PCC phones, and they just worked. If you have the relevant Cisco telephony server product you might be able to trick it into doing what you want, as that has the proper upgrader for that model of phone. I previously had experience of upgrading the Cisco build to the SIP build on Cisco 7641 handsets, which have 2 similar builds, but none of the techniques seemed to apply this time around. Cheers, Steve On Sun, 4 Dec 2016 at 16:03 Gopalakrishnan Nwrote: > Can't I upload the 3PCC firmware ? available from the Cisco website? > > Actually it came with sip88xx firmware. > > Regards . > > > On Fri, 2 Dec 2016, 10:38 p.m. Steve Davies, wrote: > > Hi, > > You have to buy the 3PCC version for this to work. Once you have this, > they work very much like the Cisco SPA handsets. > > I also ended up with a non-3PCC handset and it is useless, and as far as I > can tell they cannot be re-flashed. > > Cheers, > Steve > > > > On Fri, 2 Dec 2016 at 16:16 Gopalakrishnan N > wrote: > > Anyone tried integrating Cisco IP 8841 phone with Asterisk 11.x. I have > the phone with sip firmware came along with sip88xx-11.0.1SR xx. I tried to > upload woth TFTP due to some reason it's getting failed. Do I need to load > 3pcc firmware or anyway to Configure from the phone itself or from the > GUI? > > I have the SEPMAC.cnf.xml as well. > > Any suggestions would be appreciated. > > Regards . > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users