search for: asterisk1list

...hangup I got "Codec was alaw", but there was no improvement to the sound quality. Is there something I am doing wrong, or do I need to get in touch with our SIP trunk provider? -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk .

...lplan, a semicolon is used to denote a comment (on account of the comment mark being a valid DTMF digit). So you will have to insert a backslash before the semicolon before user=phone . -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk .

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>

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:

...dia=no ########## extensions.conf ########## [from-simwood] extension => s,1,NoOp(Call via Simwood form '${CALLERID(num)}' to '${EXTEN}') extension => s,n,Hangup() -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk .

...> protocol=tcp ; <--------------- only this line was changed. > > Presumably you have firewall rules in action. Did you enable TCP on port > 5060? > > -- > AJS > > Note: Originating address only accepts e-mail from list! If replying off- > list, change address to asterisk1list at earthshod dot co dot uk . > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.a...

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,

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:

...deal if you bought 1000 hardware SIP > phones. > (You also would probably increase your personal indispensability factor, > into > the bargain .....) > > -- > AJS > > Note: Originating address only accepts e-mail from list! If replying off- > list, change address to asterisk1list at earthshod dot co dot uk . > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.a...

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

...ext_context, next_ext, next_step exten => s,n,GoTo(${next_context},${next_ext},${next_step}) Will this work? Does Asterisk evaluate expressions like this, or does it expect literals? -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk .

...again. Just cd into the folder with your Asterisk source, run `make menuselect` and select "func_shell" (under dialplan functions). Then run `make` and finally `make install`. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _____________________________________________________________________ -- 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-u...

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

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:

...use. > > Also, remember if you have a call coming in on a PRI line and going out on > another PRI line, that's eating two of your thirty lines ..... > > -- > AJS > > Note: Originating address only accepts e-mail from list! If replying off- > list, change address to asterisk1list at earthshod dot co dot uk . > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.a...

<!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

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