Re: [asterisk-users] announcement to be played for attended
Thanks a lot Dona and jg for your inputs. I'll try to find some way to do this from Dialplan or AMI and let you guys know soon. Please share if you have some more ideas. Regards, Rajib Date: Tue, 11 Jun 2013 18:34:46 +0200 From: jg webaccou...@jgoettgens.de Subject: Re: [asterisk-users] announcement to be played for attended transfer call To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 51b751a6.5000...@jgoettgens.de Content-Type: text/plain; charset=UTF-8; format=flowed So, B transfers the call and after bridging to C, B should get an announcement. This is just an idea: See whether you can dispatch the termination of the call leg B-C by evaluating the DIALSTATUS variable. I am not sure whether you can see this inside the dialplan, but you should get the event via AMI. This is only the 1st part of the solution. A general solution would require a lot of things or may not be possible at all as you can transfer calls not only via Asterisk using DTMF signalling, but also the SIP phones themselves might be capable of transferring calls, thereby circumventing Asterisk. jg -- _ -- 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 'n Dahdi on Sun Solaris
Hello All, I am trying to install Asterisk 1.8.13.0 dahdi-complete 2.5.1 libpri 1.4.13 version. Is it possible to install dahdi on Sun Solaris? I have searched so many, but don't found any help. I am using SunOS solaris-server 5.11 11.1 i86pc i386 i86pc on Virtual Box. -- Regards, Chandrakant Solanki -- _ -- 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 'n Dahdi on Sun Solaris
On Wed, Jun 12, 2013 at 12:32:40PM +0530, Chandrakant Solanki wrote: Hello All, I am trying to install Asterisk 1.8.13.0 dahdi-complete 2.5.1 libpri 1.4.13 version. Is it possible to install dahdi on Sun Solaris? I have searched so many, but don't found any help. Maybe. But dahdi-complete you're trying to install includes dahdi-linux which is drivers for Linux. What do you need DAHDI for? I am using SunOS solaris-server 5.11 11.1 i86pc i386 i86pc on Virtual Box. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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 'n Dahdi on Sun Solaris
Actually I am trying for meetme module. On Wed, Jun 12, 2013 at 3:09 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Wed, Jun 12, 2013 at 12:32:40PM +0530, Chandrakant Solanki wrote: Hello All, I am trying to install Asterisk 1.8.13.0 dahdi-complete 2.5.1 libpri 1.4.13 version. Is it possible to install dahdi on Sun Solaris? I have searched so many, but don't found any help. Maybe. But dahdi-complete you're trying to install includes dahdi-linux which is drivers for Linux. What do you need DAHDI for? I am using SunOS solaris-server 5.11 11.1 i86pc i386 i86pc on Virtual Box. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Chandrakant Solanki -- _ -- 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 'n Dahdi on Sun Solaris
2013-06-12 11:42, Chandrakant Solanki skrev: Actually I am trying for meetme module. On Wed, Jun 12, 2013 at 3:09 PM, Tzafrir Cohen tzafrir.co...@xorcom.com mailto:tzafrir.co...@xorcom.com wrote: On Wed, Jun 12, 2013 at 12:32:40PM +0530, Chandrakant Solanki wrote: Hello All, I am trying to install Asterisk 1.8.13.0 dahdi-complete 2.5.1 libpri 1.4.13 version. Is it possible to install dahdi on Sun Solaris? I have searched so many, but don't found any help. If it's a new application you are building - Why not test asterisk 11 + confbridge? This way you won't need DAHDI. -- Johan Wilfer -- _ -- 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 'n Dahdi on Sun Solaris
There is some changes, which I made. If anybody knows ... please share knowledge for compilation of dahdi-complete and asterisk 1.8.13.0 On Wed, Jun 12, 2013 at 3:44 PM, Johan Wilfer li...@jttech.se wrote: 2013-06-12 11:42, Chandrakant Solanki skrev: Actually I am trying for meetme module. On Wed, Jun 12, 2013 at 3:09 PM, Tzafrir Cohen tzafrir.co...@xorcom.com mailto:tzafrir.cohen@xorcom.**com tzafrir.co...@xorcom.com wrote: On Wed, Jun 12, 2013 at 12:32:40PM +0530, Chandrakant Solanki wrote: Hello All, I am trying to install Asterisk 1.8.13.0 dahdi-complete 2.5.1 libpri 1.4.13 version. Is it possible to install dahdi on Sun Solaris? I have searched so many, but don't found any help. If it's a new application you are building - Why not test asterisk 11 + confbridge? This way you won't need DAHDI. -- Johan Wilfer -- __**__**_ -- 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Chandrakant Solanki -- _ -- 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] ILEC Interconnect
Hello Everyone, We are looking to interconnect with a local ILEC over an OC-n transport layer. They basically gave us two options in terms of mapping the SONET to the DS3: * VT1.5s mapping * DS1s mapping The second option is quite clear. We would MUX the connection, and plug the lines into qaud t1 cads etc... The tech mentioned that with the second option we would also need a DACS to convert back to M13 mapping. I was scared to tell him that I could not follow can someone explain that to me kindly :). I don't know much about VT1.5 mapping. Can someone kindly explain what the benefits or lack of are in choosing that option. Also what type of additional equipment we would need? In case I have overlooked something, can you gents please tell me what I will need in terms of hardware in both cases (minus routers and switches). What we are looking at is: CO | | | OC-n | v DS3 MUX | | v 21 Asterisk boxes with quad T1s Kind Regards, Nick. -- _ -- 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 sends the INTERNAL IP address of my equipments to my SIP friends?!?
Good morning, or Good afternoon! It depends :-) I have a standard Asterisk configuration: SIP friends (phones)-Asterisk-SIP gateway to PSTN converter 80.236.215.61 109.69.217.6internal IP ( 10.4.0.10/255.255.255.0) When analyzing traffic on a SIP friend/phone I see this: INVITE sip:@80.236.215.61:64946;ob SIP/2.0 Via: SIP/2.0/UDP 109.69.217.6:5060;branch=z9hG4bK52d50250;rport Max-Forwards: 70 From: sip:@109.69.217.6;tag=as15b47581 To: test sip:@109.69.217.6;tag=kp1VwHD80rA9MVdBjTF4jyFIaCkrJcjh Contact: sip:x@109.69.217.6 Call-ID: MSMhw2bsheHWAQgHlae3O7yKQ2P9EcsM CSeq: 102 INVITE User-Agent: Asterisk Require: timer Session-Expires: 1800;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 217 v=0 o=root 664087974 664087976 IN IP4 10.4.0.10 s=Asterisk c=IN IP4 10.4.0.10 t=0 0 m=audio 8652 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv My equipement IP 10.4.0.10 is visible to the user, why? Thank you, Mickael -- _ -- 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 sends the INTERNAL IP address of my equipments to my SIP friends?!?
You mean the SDP payload? You kind of need that c= is used for RTP transmission. o= always confuses me so I will just say it's important at well. You can put a proxy in the middle and do topology hiding I guess however, that is beyond the scope of this list? Kind Regards, Nick. On 6/12/13, Mickael MONSIEUR mickael.monsi...@gmail.com wrote: Good morning, or Good afternoon! It depends :-) I have a standard Asterisk configuration: SIP friends (phones)-Asterisk-SIP gateway to PSTN converter 80.236.215.61 109.69.217.6internal IP ( 10.4.0.10/255.255.255.0) When analyzing traffic on a SIP friend/phone I see this: INVITE sip:@80.236.215.61:64946;ob SIP/2.0 Via: SIP/2.0/UDP 109.69.217.6:5060;branch=z9hG4bK52d50250;rport Max-Forwards: 70 From: sip:@109.69.217.6;tag=as15b47581 To: test sip:@109.69.217.6;tag=kp1VwHD80rA9MVdBjTF4jyFIaCkrJcjh Contact: sip:x@109.69.217.6 Call-ID: MSMhw2bsheHWAQgHlae3O7yKQ2P9EcsM CSeq: 102 INVITE User-Agent: Asterisk Require: timer Session-Expires: 1800;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 217 v=0 o=root 664087974 664087976 IN IP4 10.4.0.10 s=Asterisk c=IN IP4 10.4.0.10 t=0 0 m=audio 8652 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv My equipement IP 10.4.0.10 is visible to the user, why? Thank you, Mickael -- _ -- 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] ILEC Interconnect: Basic MUX: M13 vs DCS: VT1.5 vs DS3
Hi Nick, Going from DS1 to OC-n is a multi-step process. Typically requiring a hardware device to handle each MUX step. But you can find hardware that handles multiple MUX steps together. VT1.5 is just a raw OC-n channel containing a single DS1. An M13 device converts between DS3 and DS1. A DACS (DCS or DXC) provides M13 conversion, sometimes even capable of extracting the raw VT1.5 signal directly to DS1. The ILEC transport option you choose really depends on the terminating interface. Do you want to connect with a DS3 or OC-n? No matter what hardware you choose, you will need to convert to single copper pairs (DS1/T1) to connect to your Asterisk boxes. So an M13 or DCS will be necessary to reach the DS1 level. The device you choose depends on budget and growth expectations. Typically a DCS is an expensive investment, handling hundreds of DS3's. An M13 device is typically a small unit that handles one or two DS3's. The advantage comes when you add the 29th DS1. With VT1.5 it's just adding a single channel, DS3 will require another whole DS3 to get an additional DS1. Sincerely, Brian LaVallee From: Nick Khamis sym...@gmail.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wed, 12 Jun 2013 16:19:06 -0400 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] ILEC Interconnect Hello Everyone, We are looking to interconnect with a local ILEC over an OC-n transport layer. They basically gave us two options in terms of mapping the SONET to the DS3: * VT1.5s mapping * DS1s mapping The second option is quite clear. We would MUX the connection, and plug the lines into qaud t1 cads etc... The tech mentioned that with the second option we would also need a DACS to convert back to M13 mapping. I was scared to tell him that I could not follow can someone explain that to me kindly :). I don't know much about VT1.5 mapping. Can someone kindly explain what the benefits or lack of are in choosing that option. Also what type of additional equipment we would need? In case I have overlooked something, can you gents please tell me what I will need in terms of hardware in both cases (minus routers and switches). What we are looking at is: CO | | | OC-n | v DS3 MUX | | v 21 Asterisk boxes with quad T1s Kind Regards, Nick. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ILEC Interconnect: Basic MUX: M13 vs DCS: VT1.5 vs DS3
Is there an OC-n to SIP solution that makes sense? --Don Hi Nick, Going from DS1 to OC-n is a multi-step process. Typically requiring a hardware device to handle each MUX step. But you can find hardware that handles multiple MUX steps together. VT1.5 is just a raw OC-n channel containing a single DS1. An M13 device converts between DS3 and DS1. A DACS (DCS or DXC) provides M13 conversion, sometimes even capable of extracting the raw VT1.5 signal directly to DS1. The ILEC transport option you choose really depends on the terminating interface. Do you want to connect with a DS3 or OC-n? No matter what hardware you choose, you will need to convert to single copper pairs (DS1/T1) to connect to your Asterisk boxes. So an M13 or DCS will be necessary to reach the DS1 level. The device you choose depends on budget and growth expectations. Typically a DCS is an expensive investment, handling hundreds of DS3's. An M13 device is typically a small unit that handles one or two DS3's. The advantage comes when you add the 29th DS1. With VT1.5 it's just adding a single channel, DS3 will require another whole DS3 to get an additional DS1. Sincerely, Brian LaVallee From: Nick Khamis sym...@gmail.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wed, 12 Jun 2013 16:19:06 -0400 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] ILEC Interconnect Hello Everyone, We are looking to interconnect with a local ILEC over an OC-n transport layer. They basically gave us two options in terms of mapping the SONET to the DS3: * VT1.5s mapping * DS1s mapping The second option is quite clear. We would MUX the connection, and plug the lines into qaud t1 cads etc... The tech mentioned that with the second option we would also need a DACS to convert back to M13 mapping. I was scared to tell him that I could not follow can someone explain that to me kindly :). I don't know much about VT1.5 mapping. Can someone kindly explain what the benefits or lack of are in choosing that option. Also what type of additional equipment we would need? In case I have overlooked something, can you gents please tell me what I will need in terms of hardware in both cases (minus routers and switches). What we are looking at is: CO | | | OC-n | v DS3 MUX | | v 21 Asterisk boxes with quad T1s Kind Regards, Nick. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 sends the INTERNAL IP address of my equipments to my SIP friends?!?
Mickael MONSIEUR wrote: I have a standard Asterisk configuration: SIP friends (phones) - Asterisk - SIP gateway to PSTN converter 80.236.215.61109.69.217.6 internal IP ( 10.4.0.10/255.255.255.0 ) When analyzing traffic on a SIP friend/phone I see this: INVITE sip:@80.236.215.61:64946;ob SIP/2.0 Via: SIP/2.0/UDP 109.69.217.6:5060;branch=z9hG4bK52d50250;rport Max-Forwards: 70 From: sip:@109.69.217.6 ;tag=as15b47581 To: test sip:@109.69.217.6 ;tag=kp1VwHD80rA9MVdBjTF4jyFIaCkrJcjh Contact: sip:x@109.69.217.6 Call-ID: MSMhw2bsheHWAQgHlae3O7yKQ2P9EcsM CSeq: 102 INVITE User-Agent: Asterisk Require: timer Session-Expires: 1800;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 217 v=0 o=root 664087974 664087976 IN IP4 10.4.0.10 s=Asterisk c=IN IP4 10.4.0.10 t=0 0 m=audio 8652 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv My equipement IP 10.4.0.10 is visible to the user, why? Mickael, What version of Asterisk are you running? Is the Asterisk server outside and the SIP gateway to PSTN converter inside of a NAT? What are the NAT SUPPORT and MEDIA HANDLING settings in sip.conf? Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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
