similar to: accept DMTF tone during ringing

Hi, I have asterisk installed on centos with phpagi. Also I have PRI card connect to it. it's possible to show the sip number when calling from sip number to external number thru the PRI, instead of showing the PRI number show the sip number ? Regards -Hadi.Salem

Hi, It?s possible to use sasl dovecot smtp authentication with sendmail ? Hadi.Salem

Anyone is using 1.6 in production?? Is it ready? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081107/df5bb63a/attachment.htm

I'm having problems with a new asterisk PBX install. the phones/ATAs are all linksys/cisco. They all worked before with a commercial softswitch. Most of the linksys devices offer auto, inband, INFO and AVT. I'm looking for suggestions. Thanks in advance -- One day at a time, one second if that's what it takes

Hi, Some users have been reporting a peculiar problem. The are having an issue when they dial out to some multi-level IVRs where you make 2 or 3 touchtone choices and then are connected to a live operator. When the live operator connects the operator cannot hear them or sometimes it results in dead air. With the one-way audio issue, is it possible that something has locked the channel into some

Hello, Does anyone know whether chan_sip in Asterisk supports DTMF in clock rates other than 8000? I looked for telephone-event/16000 in the changelog and in Jira but no luck. Any help would be appreciated. -- Best regards, Vlasis Chatzistavrou.

While trying to use direct_media I'm seeing RTP payload mismatches after succesful reinvites. Initial INVITE from endpoint A to asterisk has rfc4733 DMTF m=audio 35648 RTP/AVP 9 8 111 96 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 >From asterisk to upstream U: m=audio 14338 RTP/AVP 9 8 111 18 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 So the payload types in the RTP

I have three Sipura 3000 FXO untis for incoming PSTN lines on a small pbx. There is an IVR to select the extension. The DTMF tones are not being sensed so the IVR does not work and incoming calls are not being answered. I have listed my sip.conf entries. Is there any solution to this? ;Sipura units [101] type=friend host=dynamic context=default secret=mysecret mailbox=101 dtmfmode=inband

Since firmware 1.0.3.81, unless I'm imagining things, Voicemail2 seems to be having problems. The Grandstream and sip.conf were set to RFC2833 now with that setting I get extra digits during "Mailbox" and "Password" phases. 222001 instead of 2201 for example. When both are changed to "SIP info" there is no problem. But what is the new setting "DTMF Payload

Hi, Which DTMF mode do people mostly use? I've tried SIP INFO and RFC2833 but although Asterisk recognises the tones (for feature usage), the tones arent repeated to the end user. So if I call a company that has a menu system, I can't use the menu. Thanks Dan -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I'm working on an Asterisk configuration for a call center, and they bill based on the time spent talking to an agent, but not for any time spent waiting in a queue. The CDR information contains the entire duration of the call as billable seconds, including time spent waiting in the queue. I would like the billable seconds to only include the time spent actually talking to an agent.

I am in search of open-source software to track billable time for myself. A quick Google search did not find anything that is open-source but I am hoping that this group might know of such software? The platform is C7.

Hi, Would just like to ask about cdr, i have an asterisk and i would like to bill only outbound calls not extension to extension, when i'm looking at the CDR, i can't figure out which fields i need to filter all outbound calls only. e.g if i dial 00. or 9XXXXXX (for local pstn calls) those are billable, 100 101 or 102 (all local extensions) not billable. *97 for voicemail not billable,

Hi, I am requesting for a E1 connection from my telco. They are asking if I want DSS1 or SS7, and I am stuck here. Could someone tell me the difference between the two? How should I decide which one to use? Thanks in advance for your help. Mark -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Im configuring sendmail with dovecot for virtual users using password file and file for user name. on centos. dovecot-1.0.7-7.el5 sendmail-8.13.8-2.el5 dovecot ?n # 1.0.7: /etc/dovecot.conf base_dir: /var/run/dovecot/ log_path: /var/log/dovecot.log info_log_path: /var/log/dovecot.log ssl_disable: yes login_dir: /var/run/dovecot/login login_executable(default):

Hello, We are looking for commercial solution SS7 with Asterisk. It does not need to be "build-in" with Asterisk. Could anybody suggest something? Thank you in advance. Bart

Hi all, I have this project that requires me to use 8 PSTN lines and possible more. I was thinking 2 TDM cards with FXO modules. The I got to read the "Qs about FXO/FXS cards" thread and that scared me. Can anybody recommend anything that is known to work ok with no mysterious problems? I was thinking OpenSwitch12 cards. What do you guys think? Any help is appreciated. Regards, Hadi

Dear All Can you please do me favor and let me know how can I convert *.wav files into 32 bit 44 KHz ? Please be informed that I have specific sound files in *.wav format that I converted them into *.gsm format with the aid of the following command : #sox FR00003.wav FR00003.gsm It got through but the voice quality is poor . I need to convert the original *.wav sound files (their file attribute is

Dear All, I would like to ask please if someone has a AGI script that select a value from a database and dial this value as a destination number Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090220/e2aa530c/attachment.htm

Dear All Can you please do me favor and let me know how can I stop my Asterisk ? Can you please confirm if the following procedure is correct to stop it ? #/etc/init.d/asterisk stop #cd /etc/init.d #chmod 0000 asterisk Let me thank you in advance -------------- next part -------------- An HTML attachment was scrubbed... URL: