similar to: Autoreply: Re: Display IE

Hi! Thank you all for the info! But I think I haven't explained my scenario well enough. I am not relaying the calls to SIP. What happens is the following (the scenario is: a call started from an ISDN E1 PBX, through the asterisk, to the PSTN, and was answered by the remote party): 1. A CONNECT comes from the PSTN containing a Display IE (which has info sent by the telco that is used for

Hi! Thank you all for the info! But I think I haven't explained my scenario well enough. I am not relaying the calls to SIP. What happens is the following (the scenario is: a call started from an ISDN E1 PBX, through the asterisk, to the PSTN, and was answered by the remote party): 1. A CONNECT comes from the PSTN containing a Display IE (which has info sent by the telco that is used for

Hi! Thank you all for the info! But I think I haven't explained my scenario well enough. I am not relaying the calls to SIP. What happens is the following (the scenario is: a call started from an ISDN E1 PBX, through the asterisk, to the PSTN, and was answered by the remote party): 1. A CONNECT comes from the PSTN containing a Display IE (which has info sent by the telco that is used for

Hi! Thank you all for the info! But I think I haven't explained my scenario well enough. I am not relaying the calls to SIP. What happens is the following (the scenario is: a call started from an ISDN E1 PBX, through the asterisk, to the PSTN, and was answered by the remote party): 1. A CONNECT comes from the PSTN containing a Display IE (which has info sent by the telco that is used for

Hi! Thank you all for the info! But I think I haven't explained my scenario well enough. I am not relaying the calls to SIP. What happens is the following (the scenario is: a call started from an ISDN E1 PBX, through the asterisk, to the PSTN, and was answered by the remote party): 1. A CONNECT comes from the PSTN containing a Display IE (which has info sent by the telco that is used for

Hi! Thank you all for the info! But I think I haven't explained my scenario well enough. I am not relaying the calls to SIP. What happens is the following (the scenario is: a call started from an ISDN E1 PBX, through the asterisk, to the PSTN, and was answered by the remote party): 1. A CONNECT comes from the PSTN containing a Display IE (which has info sent by the telco that is used for

Hi! Thank you all for the info! But I think I haven't explained my scenario well enough. I am not relaying the calls to SIP. What happens is the following (the scenario is: a call started from an ISDN E1 PBX, through the asterisk, to the PSTN, and was answered by the remote party): 1. A CONNECT comes from the PSTN containing a Display IE (which has info sent by the telco that is used for

Hi! I am having trouble getting the following configuration to work: PHONE1 <-- rtp --> Asterisk <--IAX--> Asterisk_SRTP_1 <--- srtp ---> Asterisk_SRTP_2 <-- rtp--> PHONE2 This means, I am using regular voip clients without srtp support on both sides, but the communication between the 2 Asterisk_SRTP boxes must be secure. The Asterisk_SRTP_2 box is registered in the

Greetings, list! My boss would like some statistics on how many calls are answered out of specific queues during a given time period, and I'm not sure how exactly to obtain those stats. Here's how our queue system works. 1) Call comes in and enters our 'ring' queue where the phones ring for 15 seconds (caller hears the standard ring tone). 2) After 15 seconds, the caller

Hi guys, I don't know if this is the question ... But I have my softphones set up to use ilbc (because I found that it is better for me) and I'm trying to connect them to my provider that provides me termination through g.729. I don't have any T1 or whatever card in my server. But everytime I try to connect them they use gsm ... I don't know why ... So, that's the reason of my

Hi guys, I don't know if this is the question ... But I have my softphones set up to use ilbc (because I found that it is better for me) and I'm trying to connect them to my provider that provides me termination through g.729. I don't have any T1 or whatever card in my server. But everytime I try to connect them they use gsm ... I don't know why ... So, that's the reason of my

Hello Marco, On 7/27/07, Marco Mouta <marco.mouta at gmail.com> wrote: > > hi, > > The VoiceMail<http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2>application uses > */usr/sbin/sendmail* to mail voicemail messages to users. This can be any > sendmail-compatible MTA. In practice you can use Sendmail<http://sendmail.org/>, > Postfix

Hi guys, I don't know if this is the question ... But I have my softphones set up to use ilbc (because I found that it is better for me) and I'm trying to connect them to my provider that provides me termination through g.729. I don't have any T1 or whatever card in my server. But everytime I try to connect them they use gsm ... I don't know why ... So, that's the reason of my

Hello Marco, On 7/27/07, Marco Mouta <marco.mouta at gmail.com> wrote: > > hi, > > The VoiceMail<http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2>application uses > */usr/sbin/sendmail* to mail voicemail messages to users. This can be any > sendmail-compatible MTA. In practice you can use Sendmail<http://sendmail.org/>, > Postfix

Hi guys, I don't know if this is the question ... But I have my softphones set up to use ilbc (because I found that it is better for me) and I'm trying to connect them to my provider that provides me termination through g.729. I don't have any T1 or whatever card in my server. But everytime I try to connect them they use gsm ... I don't know why ... So, that's the reason of my

Hello Marco, On 7/27/07, Marco Mouta <marco.mouta at gmail.com> wrote: > > hi, > > The VoiceMail<http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2>application uses > */usr/sbin/sendmail* to mail voicemail messages to users. This can be any > sendmail-compatible MTA. In practice you can use Sendmail<http://sendmail.org/>, > Postfix

Hi guys, I don't know if this is the question ... But I have my softphones set up to use ilbc (because I found that it is better for me) and I'm trying to connect them to my provider that provides me termination through g.729. I don't have any T1 or whatever card in my server. But everytime I try to connect them they use gsm ... I don't know why ... So, that's the reason of my

Hello Marco, On 7/27/07, Marco Mouta <marco.mouta at gmail.com> wrote: > > hi, > > The VoiceMail<http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2>application uses > */usr/sbin/sendmail* to mail voicemail messages to users. This can be any > sendmail-compatible MTA. In practice you can use Sendmail<http://sendmail.org/>, > Postfix

Hi guys, I don't know if this is the question ... But I have my softphones set up to use ilbc (because I found that it is better for me) and I'm trying to connect them to my provider that provides me termination through g.729. I don't have any T1 or whatever card in my server. But everytime I try to connect them they use gsm ... I don't know why ... So, that's the reason of my

Hello Marco, On 7/27/07, Marco Mouta <marco.mouta at gmail.com> wrote: > > hi, > > The VoiceMail<http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2>application uses > */usr/sbin/sendmail* to mail voicemail messages to users. This can be any > sendmail-compatible MTA. In practice you can use Sendmail<http://sendmail.org/>, > Postfix