similar to: Call rating software

Hello; I am facing a trouble with A2Billing when using analogue lines because the channels are not closing properly when dialing happen through A2Billing (it seems the dialing scenario including the hangup is not handled properly through A2Billing but I do not have control on this). But when I do dialing from asterisk and using analogue lines, I do not face a trouble because I can write the

Hello all, Can someone recommend what hardware to use for a 1000 analogue line capacity asterisk PABX? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160217/2bcd322f/attachment.html>

I'm having a problem with a new SIP trunk. Calls within the UK to fixed lines are fine, but calls to mobiles have noticeably poorer audio quality. I thought it might have been a codec issue; we have used G.726 for internal and external calls (over primary ISDN and GSM). So I tried allowing "alaw", (G.711 A-law) which is the native codec used within the PSTN in this country,

On Wednesday 17 Feb 2016, imperium broadcast wrote: > I kinda have it working with chan_sip. > > Dial(SIP/+${EXTEN}\;phone-context=+44 at 10.10.10.10;user=phone) > But it doesn't include the user=phone at the end when dialling out. > > "To: <sip:+4499999999999;phone-context=+44 at 10.10.10.10>". > > even adding > usereqphone=yes > to the

Dear List wanna configure click2call in such a manner that my asterisk box call two mobile numbers and connect both numbers to talk. I have configured voip gateway, my internal and external calls are working fine. please help , abhi -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all Hi,I'm facing a strange issue where by SANGOMA not detected by goautodial system , Thats the problem : Configuring ISDN BRI cards [A500/B700] ------------------------------------ No Sangoma ISDN BRI cards detected Press any key to continue: ------------------------------------ Configuring GSM cards [W400] ------------------------------------ No Sangoma GSM cards detected

+1 spending money to get that many fxs ports is going to negate any savings of reusing analog phones instead of buying ip phones 1000 analog ports sounds like hell and if it was me I would be embarrassed to have a setup like that tied to my name if I was a consultant etc. Someone will come in after you and ask who set it up and the customer will say you :) On Feb 17, 2016 4:23 AM, "A J

Hi, Let's say I do: Set(data=xxx,yyy) Gosub(my-sub,s,1(${data})) My subroutine will only receive "xxx" for ARG1. How can I pass a literal with a comma to a single argument in a subroutine? (The point is: when calling the subroutine I do not know if the variable has a comma or not.) Thanks, Daniel -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Guys I am trying to write a macro for a call return so for example Anyone in the company transfers a call to another extension and it is not answered etc it must return to the person who did the transfer I have got it working but if the call originates externally for example someone calls in to the switchboard and they transfer it then it tries to return to the outside caller. As

Yes, it is enabled on port 5060. I do receive a TCP ACK back from the server, so I know the TCP segment is received at the server hosting the Asterisk build. On Wed, Feb 17, 2016 at 8:36 AM, A J Stiles <asterisk_list at earthshod.co.uk> wrote: > On Wednesday 17 Feb 2016, Sonny Rajagopalan wrote: > > OK. Let me ask this. Is anything else necessary, except choosing TCP as > the

Hi All, I have to forward incoming call on PRI back out to PRI but I need the original Callerid to passthrough. Is it possible with DAHDI PRI cards without involving the service provider? Thanks -- Best Ragards Rizwan H Qureshi V: +971 (0) 528272154 linkedin.com/in/rhqureshi -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi! Is possible to use X100p TDM400P, Tdm410p, Tdm400, A400p, Ax400p or any others digium card FXO for use Fax modem? Thanks.

Hi All I have an Asterisk 1.8.32.3 instance that will at random intervals stop logging CDR data to MSSQL via FreeTDS. On investigation I'll find that the FreeTDS module has been unloaded somehow. It is not listed in cdr show status or show module like. Trying module load cdr_tds results in module cdr_tds does not provide a license key and the module is not loaded. The only way to get

OK. Let me ask this. Is anything else necessary, except choosing TCP as the preferred protocol on the client, to make TCP w Asterisk work? At the moment, I have only changed one line in pjsip.conf from my working UDP setup: [transport-tcp] type=transport protocol=tcp ; <--------------- only this line was changed. On Wed, Feb 17, 2016 at 8:28 AM, Sonny Rajagopalan < sonny.rajagopalan at

Hi all, I am getting the following error when starting asterisk: pbx_functions.c: Function SHELL not registered Some of my conf files use a SHELL command, which used to work with an older version of asterisk, but now with version 13.9.1 I see this warning in the error log. How can I register the SHELL function? From what I can find in the wiki's, it should just be available? Best regards,

<!DOCTYPE html> <html><head> <meta charset="UTF-8"> </head><body><p>Hello,</p><p><br></p><p>Having installed DAHDI to be able to use the meetme() application , when I start the dahdi service it generates me the following error:</p><p>-bash: /etc/init.d/dahdi: No such file or

Thanks AJ and David, We were actually using GSM gateways by setting busy forward number on the SIMs and just giving busy signal on every incoming call, telco took care of the forwarding and the line was free within seconds. Now we need to scale up the setup but GSM gateways a very very expensive if we want to scale upto a 1000 DIDs, which means thousand SIMs and a gateway/gateways big enough.

Hi, does anyone have a recommendation for a SIP phone, which allows dialing from a phonebook, and hiding the dialed number from the end users? Also from the call history of course. It seems Mitel can do this, and I have a use case where this is a requirement. Thanks, Antonio -------------- next part -------------- An HTML attachment was scrubbed... URL: