Re: [asterisk-users] Asterisk as gateway
This is a setup of Asterisk as extension to an existing Asterisk PBX. It has to be that way and not IAX. Simply we need to an extension number with DIDs to an external PBX which is a helper to our office. This has to be done for the second PBX as well. On Thu, Mar 22, 2018 at 2:18 PM, Atux Atuxwrote: > i would like to ask how to connect 2 systems. I would like to have an > asterisk where it will have all the connections to the outside world (sip > trunks) and it will called the gateway. This asterisk will have extension > numbers of 3XX. > In the LAN there will be 2 other asterisk boxes (A & B) where A will have > the extension numbers 4XX and B the 5XX. > -gateway 3XX has all sip trunks to the outside world > -A 4XX. > -B 5XX > > I would like to have A to connect to the gateway as extension 308 and > route all calls incoming/outgoing through the gateway. the same applies to > B as extension 309. > i am kinda lost with config and the dialplan. In the gateway i have in the > sip.conf 2extensions 308 &309. in the gateway's extension.conf i have 5 > DIDs for 308 and another 5 for 309 as follows: > > 308's first DID up to 123456784 > exten => 123456780,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})}) > exten => 123456780,n,Answer() > exten => 123456780,n,Wait(1) > exten => 123456780,n,Dial(SIP/308,20) > exten => 123456780,n,VoiceMail(308@home,u) > exten => 123456780,n,Busy(3) > > > > > > 309's first DID up to 123456789 > exten => 123456785,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})}) > exten => 123456785,n,Answer() > exten => 123456785,n,Wait(1) > exten => 123456785,n,Dial(SIP/309,20) > exten => 123456785,n,VoiceMail(309@home,u) > exten => 123456785,n,Busy(3) > > > > Some help please? > John > -- _ -- 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] Asterisk as gateway
i would like to ask how to connect 2 systems. I would like to have an asterisk where it will have all the connections to the outside world (sip trunks) and it will called the gateway. This asterisk will have extension numbers of 3XX. In the LAN there will be 2 other asterisk boxes (A & B) where A will have the extension numbers 4XX and B the 5XX. -gateway 3XX has all sip trunks to the outside world -A 4XX. -B 5XX I would like to have A to connect to the gateway as extension 308 and route all calls incoming/outgoing through the gateway. the same applies to B as extension 309. i am kinda lost with config and the dialplan. In the gateway i have in the sip.conf 2extensions 308 &309. in the gateway's extension.conf i have 5 DIDs for 308 and another 5 for 309 as follows: 308's first DID up to 123456784 exten => 123456780,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})}) exten => 123456780,n,Answer() exten => 123456780,n,Wait(1) exten => 123456780,n,Dial(SIP/308,20) exten => 123456780,n,VoiceMail(308@home,u) exten => 123456780,n,Busy(3) 309's first DID up to 123456789 exten => 123456785,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})}) exten => 123456785,n,Answer() exten => 123456785,n,Wait(1) exten => 123456785,n,Dial(SIP/309,20) exten => 123456785,n,VoiceMail(309@home,u) exten => 123456785,n,Busy(3) Some help please? John -- _ -- 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] Asterisk+h324m gateway issue
Hi , i worked with h324m gateway for 3g video calling .It configured successfully . my code in extensions.conf is [from-zaptel] exten = _X.,1,h324m_gw(0@mainmenu) exten=_X.,n,Hangup [mainmenu] exten = 0,1,h324m_gw_answer() exten = 0,2,mp4play(/tmp/menu/menu.mp4,'n(1)') when i make a video call (either sip or through pri) , asterisk cli shows the following error -- Executing [123@from-zaptel:1] h324m_gw(SIP/100-b7602680, 0@mainmenu) in new stack localhost*CLI Disconnected from Asterisk server Executing last minute cleanups when i routed the call directly to [mainmenu] call stack at h324m_gw_answer() please help me ... Thanks Regards, Pankaj Pandey -- _ -- 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 cisco gateway westell isdx
Hi, I am hoping someone can help me with a problem I am having. I am trying to setup a connection from an Elastix 2 server to a Siemens isdx PBX. The setup is as follows Elastix 2 *sip trunk* Cisco 2621XM router with 2 E1 voice interfaces *QSIG* Westell IQ2000 protocol convertor *DPNSS* Siemens ISDX So the Elastix box has a SIP trunk to the cisco router which then talks QSIG to the Westell which converts it to DPNSS to talk to the ISDX. I have managed to make a call from a phone on Elastix to a phone on the ISDX but it drops after about 3 seconds, every time. Would anyone have any idea why this is? Here is the setup I have on Elastix and the Cisco router Elastix SIP trunk PEER details type=friend qualify=no nat=no insecure=very host=10.132.41.13 dtmfmode=rfc2833 context=from-internal canreinvite=yes disallow=all allow=ulaw relevant Cisco config version 12.4 service timestamps debug datetime msec service timestamps log datetime msec service password-encryption ! boot-start-marker boot-end-marker ! no network-clock-participate slot 1 no network-clock-participate wic 0 ip cef ! ! no ip domain lookup isdn switch-type primary-qsig voice-card 1 ! ! voice rtp send-recv ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco sip bind control source-interface FastEthernet0/0 bind media source-interface FastEthernet0/0 ! ! voice class codec 1 codec preference 1 g711ulaw codec preference 2 g729r8 bytes 40 ! ! controller E1 1/0 ! controller E1 1/1 pri-group timeslots 1-31 ! ! class-map match-all FAX description Match T.38 Fax match access-group name FAX ! interface FastEthernet0/0 ip address 10.132.41.13 255.255.255.0 no ip mroute-cache speed 100 full-duplex ! interface FastEthernet0/1 no ip address duplex auto speed auto ! interface Serial1/1:15 no ip address encapsulation hdlc no logging event link-status isdn switch-type primary-qsig isdn overlap-receiving isdn incoming-voice voice no cdp enable ! ip forward-protocol nd ip route 0.0.0.0 0.0.0.0 10.132.41.1 ! ip http server ! control-plane ! voice-port 1/1:15 ! ! dial-peer voice 786 voip huntstop destination-pattern 56... progress_ind setup enable 3 voice-class codec 1 session protocol sipv2 session target sip-server dtmf-relay rtp-nte no vad ! dial-peer voice 2 pots description QSIG Link to PABX via Westell destination-pattern 2 progress_ind setup enable 3 progress_ind alert enable 8 direct-inward-dial port 1/1:15 forward-digits all ! dial-peer voice 3 pots description Featurenet 7xx routing huntstop destination-pattern 7T progress_ind setup enable 3 progress_ind alert enable 8 incoming called-number 786 no digit-strip direct-inward-dial forward-digits all ! gateway ! sip-ua registrar ipv4:10.133.40.50 expires 3600 sip-server ipv4:10.133.40.50 Any help would be greatly appreciated Thanks, DT -- _ -- 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 GSM Gateway Project
Hi There http://www.2n.cz/company/2n_history.html offer this kind of products. they works vey well with asterisk. Ciao Andrea Michael Graves ha scritto: On Mon, 23 Jun 2008 10:09:21 -0400, Steve Totaro wrote: On Mon, Jun 23, 2008 at 9:57 AM, [EMAIL PROTECTED] wrote: The quad-band model is around $250 USD. See Ebay auction here http://tinyurl.com/5tvoa9 Michael Graves mgraves at mstvp.com o(713) 861-4005 c(713) 201-1262 sip:[EMAIL PROTECTED] skype mjgraves FWD 54245 Do any of these do SMS? Yes, most of them do. See #7 on the following page: http://www.portech.com.tw/eweb/index1.htm Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [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 ___ -- 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] Asterisk GSM Gateway Project
I already checked them out. If you read their fine prints well they have minutes limitations then you have to buy licenses. From the responses that I got, I can get one the pci gsm cards with the drivers and that will work for us except that it does not scale very well. On Tue, Jun 24, 2008 at 10:58 AM, Andrea Cristofanini [EMAIL PROTECTED] wrote: Hi There http://www.2n.cz/company/2n_history.html offer this kind of products. they works vey well with asterisk. Ciao Andrea Michael Graves ha scritto: On Mon, 23 Jun 2008 10:09:21 -0400, Steve Totaro wrote: On Mon, Jun 23, 2008 at 9:57 AM, [EMAIL PROTECTED] wrote: The quad-band model is around $250 USD. See Ebay auction here http://tinyurl.com/5tvoa9 Michael Graves mgraves at mstvp.com o(713) 861-4005 c(713) 201-1262 sip:[EMAIL PROTECTED][EMAIL PROTECTED] skype mjgraves FWD 54245 Do any of these do SMS? Yes, most of them do. See #7 on the following page: http://www.portech.com.tw/eweb/index1.htm Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] [EMAIL PROTECTED] skype mjgraves [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 ___ -- 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 ___ -- 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] Asterisk GSM Gateway Project
On Sun, 22 Jun 2008 08:45:10 -0500, Michael Graves wrote: I have a small Portech GSM gateway. It works well. It's GSMSIP which seems to me a better solution than FXO/FXS type interfaces. They make gateways up to 32 port for E-1 interconnect. what did they cost, michael ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ -- 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] Asterisk GSM Gateway Project
WideVOIP [EMAIL PROTECTED] writes: Junghanns have PCI cards with GSM modules and the drivers it works great They have the classic problem with interrupt sharing. At least the card we bought does. Welcome to the 21st century. /Benny ___ -- 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] Asterisk GSM Gateway Project
Never had this problem over HP or SuperMicro servers a+ Thierry -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Benny Amorsen Envoyé : lundi 23 juin 2008 08:48 À : asterisk-users@lists.digium.com Objet : Re: [asterisk-users] Asterisk GSM Gateway Project WideVOIP [EMAIL PROTECTED] writes: Junghanns have PCI cards with GSM modules and the drivers it works great They have the classic problem with interrupt sharing. At least the card we bought does. Welcome to the 21st century. /Benny ___ -- 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 ___ -- 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] Asterisk GSM Gateway Project
Please don't top post. WideVOIP [EMAIL PROTECTED] writes: Never had this problem over HP or SuperMicro servers If the card happens to not share interrupts, all is well. /Benny ___ -- 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] Asterisk GSM Gateway Project
The quad-band model is around $250 USD. See Ebay auction here http://tinyurl.com/5tvoa9 Michael Graves mgraves at mstvp.com o(713) 861-4005 c(713) 201-1262 sip:[EMAIL PROTECTED] skype mjgraves FWD 54245 Original Message Subject: Re: [asterisk-users] Asterisk GSM Gateway Project From: Dinesh Nair [EMAIL PROTECTED] Date: Mon, June 23, 2008 1:03 am To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Cc: [EMAIL PROTECTED] On Sun, 22 Jun 2008 08:45:10 -0500, Michael Graves wrote: I have a small Portech GSM gateway. It works well. It's GSMSIP which seems to me a better solution than FXO/FXS type interfaces. They make gateways up to 32 port for E-1 interconnect. what did they cost, michael ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ -- 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] Asterisk GSM Gateway Project
On Mon, Jun 23, 2008 at 9:57 AM, [EMAIL PROTECTED] wrote: The quad-band model is around $250 USD. See Ebay auction here http://tinyurl.com/5tvoa9 Michael Graves mgraves at mstvp.com o(713) 861-4005 c(713) 201-1262 sip:[EMAIL PROTECTED] skype mjgraves FWD 54245 Do any of these do SMS? Thanks, Steve T ___ -- 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] Asterisk GSM Gateway Project
On Mon, 23 Jun 2008 10:09:21 -0400, Steve Totaro wrote: On Mon, Jun 23, 2008 at 9:57 AM, [EMAIL PROTECTED] wrote: The quad-band model is around $250 USD. See Ebay auction here http://tinyurl.com/5tvoa9 Michael Graves mgraves at mstvp.com o(713) 861-4005 c(713) 201-1262 sip:[EMAIL PROTECTED] skype mjgraves FWD 54245 Do any of these do SMS? Yes, most of them do. See #7 on the following page: http://www.portech.com.tw/eweb/index1.htm Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [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] Asterisk GSM Gateway Project
Thanks all for the good feedback. On Mon, Jun 23, 2008 at 7:19 PM, Michael Graves [EMAIL PROTECTED] wrote: On Mon, 23 Jun 2008 10:09:21 -0400, Steve Totaro wrote: On Mon, Jun 23, 2008 at 9:57 AM, [EMAIL PROTECTED] wrote: The quad-band model is around $250 USD. See Ebay auction here http://tinyurl.com/5tvoa9 Michael Graves mgraves at mstvp.com o(713) 861-4005 c(713) 201-1262 sip:[EMAIL PROTECTED] [EMAIL PROTECTED] skype mjgraves FWD 54245 Do any of these do SMS? Yes, most of them do. See #7 on the following page: http://www.portech.com.tw/eweb/index1.htm Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] [EMAIL PROTECTED] skype mjgraves [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 ___ -- 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] Asterisk GSM Gateway Project
Junghanns have PCI cards with GSM modules and the drivers it works great Best Regards Thierry _ De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de broadband Voice Envoyé : samedi 21 juin 2008 23:39 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [asterisk-users] Asterisk GSM Gateway Project I am thinking about using an existing asterisk box and turning it into a gsm gateway. Has anyone tried this before, adding sonme gsm cards and an antenna. Any ideas. ___ -- 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] Asterisk GSM Gateway Project
I have a small Portech GSM gateway. It works well. It's GSMSIP which seems to me a better solution than FXO/FXS type interfaces. They make gateways up to 32 port for E-1 interconnect. Michael --Original Message Text--- From: broadband Voice Date: Sat, 21 Jun 2008 17:38:50 -0400 I am thinking about using an existing asterisk box and turning it into a gsm gateway. Has anyone tried this before, adding sonme gsm cards and an antenna. Any ideas. -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [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] Asterisk GSM Gateway Project
I posted at almost the exact moment you did Michael! Taking a break from the move. On Sun, Jun 22, 2008 at 3:45 PM, Michael Graves [EMAIL PROTECTED] wrote: I have a small Portech GSM gateway. It works well. It's GSMSIP which seems to me a better solution than FXO/FXS type interfaces. They make gateways up to 32 port for E-1 interconnect. Michael --Original Message Text--- From: broadband Voice Date: Sat, 21 Jun 2008 17:38:50 -0400 I am thinking about using an existing asterisk box and turning it into a gsm gateway. Has anyone tried this before, adding sonme gsm cards and an antenna. Any ideas. -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [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 ___ -- 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] Asterisk GSM Gateway Project
Take a look here http://www.smallnetbuilder.com/content/view/30428/82/ Michael Graves wrote a great 2 part article about this. On Sun, Jun 22, 2008 at 2:13 PM, WideVOIP [EMAIL PROTECTED] wrote: Junghanns have PCI cards with GSM modules and the drivers it works great Best Regards Thierry De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de broadband Voice Envoyé : samedi 21 juin 2008 23:39 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [asterisk-users] Asterisk GSM Gateway Project I am thinking about using an existing asterisk box and turning it into a gsm gateway. Has anyone tried this before, adding sonme gsm cards and an antenna. Any ideas. ___ -- 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 ___ -- 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] Asterisk GSM Gateway Project
When you say GSM gateway, do you mean using cell phones as FXOs or do you mean using Asterisk coupled with a BTS and having your own little cell? Thanks, Steve Totaro On Sun, Jun 22, 2008 at 8:13 AM, WideVOIP [EMAIL PROTECTED] wrote: Junghanns have PCI cards with GSM modules and the drivers it works great Best Regards Thierry De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de broadband Voice Envoyé : samedi 21 juin 2008 23:39 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [asterisk-users] Asterisk GSM Gateway Project I am thinking about using an existing asterisk box and turning it into a gsm gateway. Has anyone tried this before, adding sonme gsm cards and an antenna. Any ideas. ___ -- 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 ___ -- 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] Asterisk GSM Gateway Project
On Sun, 22 Jun 2008, Michael Graves wrote: I have a small Portech GSM gateway. It works well. It's GSMSIP which seems to me a better solution than FXO/FXS type interfaces. They make gateways up to 32 port for E-1 interconnect. I've used a 2-channel one too. They also do 4-channels ones - relatively economical too - compared to PCI cards, and as they're Ethernet/SIP based then in-theory you can put dozens of them online with no need for drivers, or PCI slots, etc. Gordon ___ -- 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] Asterisk GSM Gateway Project
On Sun, 22 Jun 2008 18:08:04 +0100 (BST), Gordon Henderson wrote: On Sun, 22 Jun 2008, Michael Graves wrote: I have a small Portech GSM gateway. It works well. It's GSMSIP which seems to me a better solution than FXO/FXS type interfaces. They make gateways up to 32 port for E-1 interconnect. I've used a 2-channel one too. They also do 4-channels ones - relatively economical too - compared to PCI cards, and as they're Ethernet/SIP based then in-theory you can put dozens of them online with no need for drivers, or PCI slots, etc. Gordon Actually, Portech make 8 port units already, and much larger ones. The 1, 2, 4 8 port units are all based on the same hardware, just with multiple instances of it in the same chassis. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [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
[asterisk-users] Asterisk GSM Gateway Project
I am thinking about using an existing asterisk box and turning it into a gsm gateway. Has anyone tried this before, adding sonme gsm cards and an antenna. Any ideas. ___ -- 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] Asterisk GSM Gateway Project
There is a commercial product that does just that. I cannot reveal the company name since they are clients of mine but they have a BTS, a retractable 15 foot tower, a laptop or small PC running Asterisk and either do VoIP over VSAT or connect via T1/E1. Mostly government work but they are busy and growing very fast. Thanks, Steve Totaro On Sat, Jun 21, 2008 at 5:38 PM, broadband Voice [EMAIL PROTECTED] wrote: I am thinking about using an existing asterisk box and turning it into a gsm gateway. Has anyone tried this before, adding sonme gsm cards and an antenna. Any ideas. ___ -- 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 ___ -- 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] Asterisk GSM Gateway Project
Google multitech CallFinder 100, has both FXO and FXS interface you can connect. Problem is you call out and don't get simulated ring back while the GSM call is being setup. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Saturday, June 21, 2008 11:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk GSM Gateway Project There is a commercial product that does just that. I cannot reveal the company name since they are clients of mine but they have a BTS, a retractable 15 foot tower, a laptop or small PC running Asterisk and either do VoIP over VSAT or connect via T1/E1. Mostly government work but they are busy and growing very fast. Thanks, Steve Totaro On Sat, Jun 21, 2008 at 5:38 PM, broadband Voice [EMAIL PROTECTED] wrote: I am thinking about using an existing asterisk box and turning it into a gsm gateway. Has anyone tried this before, adding sonme gsm cards and an antenna. Any ideas. ___ -- 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 ___ -- 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 - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- 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
[asterisk-users] Asterisk: SIP Gateway or Proxy
Hi, I realise this might be an insane noob question, but I'm on a huge brain freeze, and I'm trying to decide this: Is Asterisk a SIP Gateway or SIP proxy? -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Asterisk: SIP Gateway or Proxy
yusuf wrote: Hi, I realise this might be an insane noob question, but I'm on a huge brain freeze, and I'm trying to decide this: Is Asterisk a SIP Gateway or SIP proxy? Short answer: Gateway. This has been discussed to death many times on this list. Please search the archive for more details. Leo ___ --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] Asterisk: SIP Gateway or Proxy
Asterisk can actually act as a Gateway and a SIP Proxy. This is where a lot of confusion comes in. It can do pretty much any voip function you throw at it. Definitely search the archives if you still have questions. try site:lists.digium.com keyword in google to search the mail archives. David On 12/1/06, yusuf [EMAIL PROTECTED] wrote: Hi, I realise this might be an insane noob question, but I'm on a huge brain freeze, and I'm trying to decide this: Is Asterisk a SIP Gateway or SIP proxy? -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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
Re: [asterisk-users] Asterisk: SIP Gateway or Proxy
Not a proxy, of course, or proxy for a very small amount of users. Pbx and gateway is the best usage. David Thomas a écrit : Asterisk can actually act as a Gateway and a SIP Proxy. This is where a lot of confusion comes in. It can do pretty much any voip function you throw at it. Definitely search the archives if you still have questions. try site:lists.digium.com keyword in google to search the mail archives. David On 12/1/06, yusuf [EMAIL PROTECTED] wrote: Hi, I realise this might be an insane noob question, but I'm on a huge brain freeze, and I'm trying to decide this: Is Asterisk a SIP Gateway or SIP proxy? -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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 ___ --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] Asterisk: SIP Gateway or Proxy
On 01/12/06, yusuf [EMAIL PROTECTED] wrote: Hi, I realise this might be an insane noob question, but I'm on a huge brain freeze, and I'm trying to decide this: Is Asterisk a SIP Gateway or SIP proxy? http://www.voip-info.org/wiki-Asterisk+SIP+not-proxy -- Peter Bowyer 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 Media Gateway
I am planning to use asterisk with Digium TDM2404E card as a media gateway to terminate traffic to Cell phones. Anyone got this working before with no problmes, specially with Answer/Disconnect supervision? Thanks ___ --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 SIP Gateway behind NATS - SIP/2.0 404 Not Found
Hi all, I am running an Asterisk server behind a NAT. I want to forward the calls from PSTN to a SIP phone (no nat and also an asterisk). I set the externip and localnet in sip.conf already. I opened the ports in my firewall. (I changed SIP port from 5060 to 5065 and limited the rtp port to 12000-13000) However, I just can't call out. I've always received SIP/2.0 404 Not Found. My sip.conf looks somewhat like this [general] context=default; Default context for incoming calls bindport=5065; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes; Enable DNS SRV lookups on outbound calls externip = 83.xxx.xxx.xxx; Address that we're going to put in outbound SIP messages localnet=10.2.70.0/255.255.255.0 localnet=192.168.18.0/255.255.255.0; All RFC 1918 addresses are local networks [thephone] type=peer host=thephonedomain.com port=5065 username=abcd nat=no usereqphone = yes ;canreinvite=no If I made a call to local SIP phone, it works fine. But to the SIP phone outside the NAT, it just doesn't seem to work. I have no idea what else I should do. Anybody could give me some suggestion?? regards, Pim ___ --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] Asterisk and gateway
Hi ok then just add the same in to sip.conf and same config made change in Extension.conf for outboud routing ram On 3/22/06, Mohammad Salaque [EMAIL PROTECTED] wrote: Dear Ram .u miss something . as i told u . my provider didn't give me anyusername /passwd. they just give me IP address .as they are gateway provider not gatekeeper .i need to send my traffic to there IP address.they give me only IPaddress nothing elsethanksSalaqueOn 3/22/06, ram [EMAIL PROTECTED] wrote: Hi ya thats correct [voip.provider.net-out] type=peer secret=password username=2345 host=ipaddress fromuser=2345 nat=yes In extensions.conf you'd then use a statement like this: exten = _9.,1,Dial( SIP/${EXTEN:[EMAIL PROTECTED],30,r) just example above should help you ram On 3/22/06, Mohammad Salaque [EMAIL PROTECTED] wrote: thanks ram all they give me an ip address. and told me to send SIP traffic. soon sip.conf should i add only these ? [worlgateay] host= xxx.xxx.xxx.xxx thanks Salaque On 3/22/06, ram [EMAIL PROTECTED] wrote: Hi add that information ( which you got from SIP provider) in to sip.conf and make changes according in extension.conf for routing. ram On 3/22/06, Mohammad Salaque [EMAIL PROTECTED] wrote: Dear all, I am really new in this world, In my office i setup a Asterisk and all extensions are working fine . i also add some trunk to route my local calls. As some user needs to call overseas so we are looking for cheap gatewaywith SIP supported. Now . we found one gateway . they just give me there softwitch's ip and told to me to send my call there. i have no idea how to router my traffic from my Asterisk box to that ip . as from all other service i got a username/passwd . Could anyone give me any idea ? thanks ___ --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 -- Got my mail ? ___ --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--Got my mail ?___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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
[Asterisk-Users] Asterisk and gateway
Dear all, I am really new in this world, In my office i setup a Asterisk and all extensions are working fine . i also add some trunk to route my local calls. As some user needs to call overseas so we are looking for cheap gateway with SIP supported. Now . we found one gateway . they just give me there softwitch's ip and told to me to send my call there. i have no idea how to router my traffic from my Asterisk box to that ip . as from all other service i got a username/passwd . Could anyone give me any idea ? thanks ___ --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] Asterisk and gateway
Hi add that information ( which you got from SIP provider) in to sip.conf and make changes accordingin extension.conf for routing. ram On 3/22/06, Mohammad Salaque [EMAIL PROTECTED] wrote: Dear all,I am really new in this world, In my office i setup a Asterisk and allextensions are working fine . i also add some trunk to route my local calls. As some user needs to call overseas so we are looking for cheapgatewaywith SIP supported.Now . we found one gateway . they just give me there softwitch's ipand told to me to send my call there. i have no idea how to router my traffic from my Asterisk box to thatip . as from all other service i got a username/passwd .Could anyone give me any idea ?thanks___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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
Re: [Asterisk-Users] Asterisk and gateway
thanks ram all they give me an ip address. and told me to send SIP traffic. so on sip.conf should i add only these ? [worlgateay] host= xxx.xxx.xxx.xxx thanks Salaque On 3/22/06, ram [EMAIL PROTECTED] wrote: Hi add that information ( which you got from SIP provider) in to sip.conf and make changes according in extension.conf for routing. ram On 3/22/06, Mohammad Salaque [EMAIL PROTECTED] wrote: Dear all, I am really new in this world, In my office i setup a Asterisk and all extensions are working fine . i also add some trunk to route my local calls. As some user needs to call overseas so we are looking for cheap gateway with SIP supported. Now . we found one gateway . they just give me there softwitch's ip and told to me to send my call there. i have no idea how to router my traffic from my Asterisk box to that ip . as from all other service i got a username/passwd . Could anyone give me any idea ? thanks ___ --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 -- Got my mail ? ___ --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] Asterisk and gateway
Hi ya thats correct [voip.provider.net-out]type=peersecret=passwordusername=2345host=ipaddressfromuser=2345nat=yesIn extensions.conf you'd then use a statement like this: exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r) just example above should help you ram On 3/22/06, Mohammad Salaque [EMAIL PROTECTED] wrote: thanks ramall they give me an ip address. and told me to send SIP traffic.soon sip.conf should i add only these ? [worlgateay]host= xxx.xxx.xxx.xxxthanksSalaqueOn 3/22/06, ram [EMAIL PROTECTED] wrote: Hi add that information ( which you got from SIP provider) in to sip.conf and make changes according inextension.conf for routing. ram On 3/22/06, Mohammad Salaque [EMAIL PROTECTED] wrote: Dear all, I am really new in this world, In my office i setup a Asterisk and all extensions are working fine . i also add some trunk to route my local calls. As some user needs to call overseas so we are looking for cheap gatewaywith SIP supported. Now . we found one gateway . they just give me there softwitch's ip and told to me to send my call there. i have no idea how to router my traffic from my Asterisk box to that ip . as from all other service i got a username/passwd . Could anyone give me any idea ? thanks ___ --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 --Got my mail ?___--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
Re: [Asterisk-Users] Asterisk and gateway
Dear Ram . u miss something . as i told u . my provider didn't give me any username /passwd. they just give me IP address . as they are gateway provider not gatekeeper . i need to send my traffic to there IP address. they give me only IP address nothing else thanks Salaque On 3/22/06, ram [EMAIL PROTECTED] wrote: Hi ya thats correct [voip.provider.net-out] type=peer secret=password username=2345 host=ipaddress fromuser=2345 nat=yes In extensions.conf you'd then use a statement like this: exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r) just example above should help you ram On 3/22/06, Mohammad Salaque [EMAIL PROTECTED] wrote: thanks ram all they give me an ip address. and told me to send SIP traffic. so on sip.conf should i add only these ? [worlgateay] host= xxx.xxx.xxx.xxx thanks Salaque On 3/22/06, ram [EMAIL PROTECTED] wrote: Hi add that information ( which you got from SIP provider) in to sip.conf and make changes according in extension.conf for routing. ram On 3/22/06, Mohammad Salaque [EMAIL PROTECTED] wrote: Dear all, I am really new in this world, In my office i setup a Asterisk and all extensions are working fine . i also add some trunk to route my local calls. As some user needs to call overseas so we are looking for cheap gateway with SIP supported. Now . we found one gateway . they just give me there softwitch's ip and told to me to send my call there. i have no idea how to router my traffic from my Asterisk box to that ip . as from all other service i got a username/passwd . Could anyone give me any idea ? thanks ___ --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 -- Got my mail ? ___ --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 -- Got my mail ? ___ --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 as Gateway
Hello to all I succefully installed Asterisk and an Eicon Diva Server 4 BRI (with CAPI) to connect to a Siemens PBX, but I still cant forward calls to the Siemens PBX (neither receive them from the PBX). Here s the result in the asterisk console when I try to dial the 116 PBX phone: -- Executing Dial(SIP/193.136.2.205:5060-fd1f, CAPI/12345678:b116|90) in new stack -- data = 12345678:b116 -- capi request omsn = 12345678 == found capi with omsn = 12345678 == CAPI Call CAPI[contr1/12345678]/11 with B3-- Called 12345678:b116 -- CONNECT_CONF ID=001 #0x0012 LEN=0014 Controller/PLCI/NCCI= 0x301 Info= 0x0 -- CONNECT_CONF ID=001 #0x0012 LEN=0014 Controller/PLCI/NCCI= 0x301 Info= 0x0 == received CONNECT_CONF PLCI = 0x301 INFO = 0 -- DISCONNECT_IND ID=001 #0x001b LEN=0014 Controller/PLCI/NCCI= 0x301 Reason = 0x3302 == DISCONNECT_IND PLCI=0x301 REASON=0x3302 -- CAPI Hangingup == No one is available to answer at this time this is my CAPI.CONF ; CAPI config ; ; [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] msn=12345678 incomingmsn=* controller=1 softdtmf=1 accountcode= context=demo ;echosquelch=1 ;echocancel=yes ;echotail=64 ;callgroup=1 ;deflect=12345678 devices=2 -this is my EXTENSIONS.CONF [from-sip] exten = _XXX,1,Dial,CAPI/12345678:b${EXTEN}|90 Does someone have an ideia of what is missing? The Siemens PBX should forward the call to its 116 extension... but there's no way I can debug it... Joao Pereira ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as Gateway
On Mon, 11 Jul 2005, Joao Pereira wrote: Hello to all I succefully installed Asterisk and an Eicon Diva Server 4 BRI (with CAPI) to connect to a Siemens PBX, but I still cant forward calls to the Siemens PBX (neither receive them from the PBX). Here s the result in the asterisk console when I try to dial the 116 PBX phone: -- Executing Dial(SIP/193.136.2.205:5060-fd1f, CAPI/12345678:b116|90) in new stack -- data = 12345678:b116 -- capi request omsn = 12345678 == found capi with omsn = 12345678 == CAPI Call CAPI[contr1/12345678]/11 with B3-- Called 12345678:b116 -- CONNECT_CONF ID=001 #0x0012 LEN=0014 Controller/PLCI/NCCI= 0x301 Info= 0x0 -- CONNECT_CONF ID=001 #0x0012 LEN=0014 Controller/PLCI/NCCI= 0x301 Info= 0x0 == received CONNECT_CONF PLCI = 0x301 INFO = 0 -- DISCONNECT_IND ID=001 #0x001b LEN=0014 Controller/PLCI/NCCI= 0x301 Reason = 0x3302 == DISCONNECT_IND PLCI=0x301 REASON=0x3302 -- CAPI Hangingup == No one is available to answer at this time Does someone have an ideia of what is missing? The Siemens PBX should forward the call to its 116 extension... but there's no way I can debug it... I assume you use chan_capi-0.3.5 !? Some messages are missing in the debug, please try chan_capi-0.5.3 from sourceforge. (But note, the capi.conf and dial syntax has changed in that version). Armin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Subject: [Asterisk-Users] asterisk gsm gateway hardware
Hello, I would like to implement a home GSM gateway using asterisk. What would you recommend me as a low-cost hardware for creating a gsm channel? I found 2n gsm gateway, that supports sip and chan_blue for bluetooth connections. Any recommendations? Basically, I want to end calls to some GSM number in my sip telephone and for some prefixes dial out using that same sip telephone. Also sending and receiving SMS will be a plus. I have a friend living in luxembourg, which would like a slovak phone number to communicate with friends. It would end on my server at home and all calls to his sim card will be routed to his ip telephone in luxembourg (and vice versa). Support for more than one sim card is a plus. Since it's a home/hobby use, I would prefer a low-cost solution. Any ideas (may be off-list) are welcome). The solution I use works very well if you need to be able to take the mobile phone away with you when you leave the house. I use a phonelabs.com dock'n'talk with a bluetooth module connected to a standard digium one-port FXO card (XP100). I can make and receive GSM calls via my mobile from asterisk, treating it as just another channel. The phone automatically connects to the dock'n'talk when it comes into bluetooth range. If you are happy with a fixed solution, where you leave the SIM permanently installed, you might want to look for a Nokia Premicell or equivalent on e-bay. This would also connect to a standard FXO port. HTH Patrick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk gsm gateway hardware recommendation?
Hello, I would like to implement a home GSM gateway using asterisk. What would you recommend me as a low-cost hardware for creating a gsm channel? I found 2n gsm gateway, that supports sip and chan_blue for bluetooth connections. Any recommendations? Basically, I want to end calls to some GSM number in my sip telephone and for some prefixes dial out using that same sip telephone. Also sending and receiving SMS will be a plus. I have a friend living in luxembourg, which would like a slovak phone number to communicate with friends. It would end on my server at home and all calls to his sim card will be routed to his ip telephone in luxembourg (and vice versa). Support for more than one sim card is a plus. Since it's a home/hobby use, I would prefer a low-cost solution. Any ideas (may be off-list) are welcome). Thanks, Juraj. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk as gateway with oh323 channel to VOIP provider that can provide gateway or gatekeeper feature ?
Hi, sorry for my h323 dumbness. VOIP provider terminates H323 calls - it can be used as gatekeeper or gateway (they claim so). What option and what setup is best to connect Asterisk to this provider ? Any working examples ? Thanks in advance, regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk SIP gateway -- SCCP Phone
I have cisco phones running SCCP, and a cisco 2600 with FXO Im using for PSTN access. I can dial out, but inbound calls are not ringing a phone. Please see my config In the 2600 Im PLARing the line and I have a SIP Dial-Peer for 4001 voice-port 1/1/0 output attenuation 0 echo-cancel coverage 32 no comfort-noise timing hookflash-out 50 connection plar opx 4001 dial-peer voice 2 voip description Route calls starting with 4 to the Asterisk PBX destination-pattern 4... session protocol sipv2 session target ipv4:10.0.50.150:5060 session transport udp dtmf-relay rtp-nte codec g711ulaw no vad In Asterisk I do get a SIP READ on inbound call --- Sip read: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.50.1:5060 From: sip:10.0.50.1;tag=C6D516C-2308 To: sip:[EMAIL PROTECTED];tag=as75c5e8e9 Date: Wed, 15 Sep 2004 14:30:22 GMT Call-ID: [EMAIL PROTECTED] Max-Forwards: 6 Content-Length: 0 CSeq: 101 ACK -- But thats all I get, no phone ring The phone is SCCP.. and the phones CAN call each other. Ideas? Justin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk NAT Gateway Setup
I am currently using Asterisk behind Belkin NAT router. With what ever NAT router I have used, I have had difficulties in registration and audio problems with my SIP provider (Iconnect and Nikotel) It was suggested that I try to connect the asterisk box directly to the internet avoiding the NAT transition. As I will still need internet connectivity, I am trying to make the asterisk box the NAT gateway. I have an additional NIC for my Asterisk box. As I am no Linux or Asterisk expert, can anyone make suggestions as to this approach and any recommended steps to accomplish this? Also, how would Asterisk know which interface to bind to? I know there is a bindaddress= parameter in the SIP config, but the address to the internet is dynamic via DHCP from my cable provider. Thanks, Kevin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk to gateway
Is it possible to send a call from the asterisk server to a gateway via sipv2 protocol. I have some 7960 phones that can receive a call from a 5350 via sipv2 and the phone can send to the gateway via sipv2. Is there an exten that dials to a gateways ? Michael K. Rodriguez DialMex LLC NOC Engineer 200 S. 10th Street Suite 1209 McAllen, TX 78501 (956) 994-0014 x107 office (956) 239-0627 mobile (956) 682-5821 fax [EMAIL PROTECTED] Escalation Procedure +++The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and destroy any copies of this document.+++