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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

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

Dear all, I am analysing growth data - response variable - using GAM and GAMM models, and 4 covariates: mean size, mean capture year, growth interval, having tumors vs. not The models work fine, and fit the data well, however when I try to compare models using AIC I cannot get an AIC value. This is the code for the gam model:

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