[asterisk-users] voicemails and recordings have words repeated

Hi all i have run into a problem and cant seem to find the solution calls that are recorded and lots of voicemails recorded you can her some of the words repeated as if the person has said it twice it happens by different callers using pjsip on 18.9.0 any ideas? thanks --

[asterisk-users] How to escape the & in BackGround (Dovid Bender

On Mon, Jan 17, 2022, 19:58 wrote: > Send asterisk-users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with

[asterisk-users] fax settings for audiocodes mp series fxs gateway and asterisk

Hi all Does anyone have working settings for a audiocodes fxs gateway behind a firewall to send faxes thru asterisk not behind nat i have tried multiple settings and haven't gotten it to work even partially thanks, israel -- _

[asterisk-users] please update contrib/scripts/get_mp3_source.sh to use https

contrib/scripts/get_mp3_source.sh svn export has to be changed to https else it fails thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at:

Re: [asterisk-users] DTMF rfc2833 missed when transfering to another server

well looks likes we solved it the rtpkeepalive was set to 5 seconds on the trunk and every time asterisk sends a rtpkeepalive a cn packet is sent the same time a cn packet is sent asterisk loses the dtmf it was sent On Wed, Dec 16, 2020 at 7:43 PM Israel Gottlieb wrote: > Hi all >

[asterisk-users] DTMF rfc2833 missed when transfering to another server

Hi all i have a asterisk server 16.11.1 (server A) that gets a call (leg A) and then calls a second server (leg B) server B is a freeswitch server the servers are configured all thru with rfc2833 for dtmf the caller enters a number a long 15 digit number like a credit card number or even a phone

[asterisk-users] Fwd: blf problems after dialplan reload

Hi Guys we have a system that uses a lot of custom hints based on the extension the extensions use the format of ext-system for example 200-pbx01 when starting asterisk the "core show hints" show the correct hints and blf works as expected in the extensions.conf we have _.,hint,Custom:${exten}

Re: [asterisk-users] i sided recordings in asterisk 16.10

-28780 i do see alot of asterisk notices in asterisk 16 alot translate.c: 12547 lost frame(s) 12548/0 (slin@8000)->(alaw@8000) On Tuesday 12 May 2020 at 12:28:51, Israel Gottlieb wrote: >* Hi guys i upgraded to asterisk 16.10 * >From what? Did you change anything else at the

[asterisk-users] i sided recordings in asterisk 16.10

Hi guys i upgraded to asterisk 16.10 and in most recordings you here only leg A in the recording sometimes you might hear a word of leg B Did any body hit this problem? Thanks, israel -- _ -- Bandwidth and Colocation Provided by

[asterisk-users] no video when dialing between extension

hi all im trying to call a door phone supporting video i hear the audio but dont get video i see this in the log why should it try to translate? Unable to find a codec translation path: (h264) -> (opus) asterisk version 13.26 thanks for any help --

[asterisk-users] codec opus on centos 6 with asterisk 16

Hi list does anyone know how i could use codec opus with asterisk 16 when using centos 6 the prebuilt binary from digium doesnt load Thanks, Israel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com --

Re: [asterisk-users] Wanted: professional softphone

look at zoiper oem.zoiper.com you could create a url that creates a build with all credentials -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at:

[asterisk-users] bridging in a 3rd caller without putting a caller on hold

hi all How could conferance in a 3rd caller without put the second caller on hold i would like to press a feature code mid call and have a 3rd caller enter the call this could be a real person or a automated system to take credit card info mid call thanks, Israel --

Re: [asterisk-users] asterisk-users Digest, Vol 179, Issue 1

how about sticking in a pbx between [c] and [h] so when [h] hangsup you send to [s] if that is 3rd party else i dont see how you could redirect [c] at all else maybe ask them to have [h] redirect [c] to [s] then [h] will also be out of the call On Mon, Jul 1, 2019, 20:03 Send asterisk-users

Re: [asterisk-users] Voicemail asking for login

Does he have the same voicemail context?

Re: [asterisk-users] multiple outbound invites

Maybe your firewall is blocking receiving packets from that provider or some sip helper is messing the returning packets so asterisk is not recieving a response and resending the invite   Original Message   From: j...@jeff.net Sent: February 22, 2017 7:57 PM To: asterisk-users@lists.digium.com

Re: [asterisk-users] First SIP-registering succeeds, second doesn't

Disable all sip alg/helpers in the router   Original Message   From: andregronwal...@gmail.com Sent: February 13, 2017 6:40 PM To: asterisk-users@lists.digium.com Reply-to: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] First SIP-registering succeeds, second doesn't Some further

Re: [asterisk-users] Call List Campaign to an IVR

there are providers which let you call directly to voicemail by using a prefix On Mon, Feb 6, 2017 at 8:28 PM, Tech Support wrote: > I remember doing the testing and two calls going out at the same time > don’t actually have to go out at the *exact* same time. The

Re: [asterisk-users] How to send SIP_NOTIFY messages with variable content ?

snom could get lots of configuration options thru sip notify i once tried updateing the display name on hot desking but ran in to his problem of having to add it to sip conf staticly On Wed, Jan 18, 2017 at 5:13 PM, Mark Wiater wrote: > > On 1/18/2017 9:58 AM, Tech

Re: [asterisk-users] Asterisk 11.24.1 garbled audio

Why not just timing test It shows the timer used On Nov 16, 2016 8:13 AM, "Stefan Viljoen" wrote: > Date: Tue, 15 Nov 2016 17:52:07 +0100 > From: Olivier > To: Asterisk Users Mailing List - Non-Commercial Discussion >

Re: [asterisk-users] Dial and start music on hold after timeout

Are you sending progress? בתאריך 24 באוג׳ 2016 13:40,‏ "Saint Michael" כתב: > ​I have the same exact issue. I cannot push any sounds or even Playtones > to the caller, unless the channel is answered, which is not possible for > billing reasons. > I am also using the Local

Re: [asterisk-users] Dial and start music on hold after timeout

e Playback means that SIP/alice should continue > to ring for the remaining 20 of the 40 seconds, as the Playback will not > answer (terminate) the call. > > Don't forget AstriCon this year - www.astricon.net > > On 23 August 2016 at 12:52, Israel Gottlieb <isr...@gmail.com&g

Re: [asterisk-users] Dial and start music on hold after timeout

You could m and make a moh file that has ringing the first 30 sec and then the anouncment בתאריך 22 באוג׳ 2016 7:19 PM,‏ "Jean Aunis" כתב: > Thank you for the idea. The problem with RetryDial, is that it will cancel > the first call, play the announce and then dial the

Re: [asterisk-users] Asterisk & Vitelity Invite issues

Could you please write the problem your having and not the reason to the problem Maybe the reason is something else בתאריך 8 באוג׳ 2016 17:25,‏ "Tammy Firefly" כתב: Hi All, We have asterisk 11.23 running sip to vitelity and from there IAX trunks split off to where they

Re: [asterisk-users] Original Callerid on transfer in asterisk 13

attempting to view the original CallerId? > > Matthew Fredrickson > > On Wed, Aug 10, 2016 at 2:59 PM, Israel Gottlieb <isr...@gmail.com> wrote: > > Hi > > Is there any configuration change in asterisk 13.9.1 to show original > > callerid on a transfer &g

[asterisk-users] Original Callerid on transfer in asterisk 13

Hi Is there any configuration change in asterisk 13.9.1 to show original callerid on a transfer In asterisk 11.21 it works as expected Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to

Re: [asterisk-users] Removing mailbox and password prompt for voicemail

Hi Could please show us your dialplan as you have it now and the lines in the log on that call it would probably help a lot more בתאריך 4 באוג׳ 2016 5:00 PM,‏ "Nabeel" כתב: > What happens when you dial "*98" from your own >> phone. > > > I get password prompt if a

Re: [asterisk-users] how to read sip debug

Another nice sip packet is sngrep Shows realtime the sip flows But i think you have to chk the asterisk answer in the dialplan logic to chk what context its hitting etc. בתאריך 6 ביולי 2016 10:05 PM,‏ "Steve Edwards" כתב: > On Wed, 6 Jul 2016, Victor Villarreal wrote:

Re: [asterisk-users] Delay after Answer

Another thing i would check is encryption is disabled on the snom בתאריך 8 ביוני 2016 10:07,‏ "Israel Gottlieb" <isr...@gmail.com> כתב: > Are you using stun? I have seen that when using stun > בתאריך 8 ביוני 2016 09:54,‏ "Faheem Muhammad" <faheem2...@gmai

Re: [asterisk-users] Delay after Answer

Are you using stun? I have seen that when using stun בתאריך 8 ביוני 2016 09:54,‏ "Faheem Muhammad" כתב: > > > Are you sure *nslookup *command is returning as expected? > Also check the output of the below command. > >> hostname && hostname -s && hostname -f > > > On Tue,

Re: [asterisk-users] variable to get waittime of caller exiting queue

e1} -${calltime}]) > exten=_X.,n,NoOp(diff) > > - > > Regards, > Muhammad > > > On Wed, May 18, 2016 at 5:05 PM, Israel Gottlieb <isr...@gmail.com> wrote: > >> Hi all >> >> Is there anywa

[asterisk-users] variable to get waittime of caller exiting queue

Hi all Is there anyway i could get in the dialplan the amount of time a caller waited in the queue before exiting? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for

Re: [asterisk-users] How is Queue avg holdtime and avg talktime calculated

alik <i...@pack-net.co.uk> wrote: > >> >> >> On 11 May 2016 at 10:24, Israel Gottlieb <isr...@gmail.com> wrote: >> >>> >>> Hi all >>> >>> How is avg hold time and avg talktime calculated and over long a period >>>

[asterisk-users] How is Queue avg holdtime and avg talktime calculated

Hi allHow is avg hold time and avg talktime calculated and over long a period of time?Thanks,Israel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory

Re: [asterisk-users] Voice recognition IVR Is it possible?

if i remember correctly nerdvittles has a article "google speech recognition api asterisk" brings results On Tue, Feb 23, 2016 at 11:56 PM, Frank wrote: > On Tue, 2016-02-23 at 17:06 +, Steve Howes wrote: > > > Google?... > > Yeah... searched "google voice

[asterisk-users] תשובה: Dialing a call back out on same SIP trunk as it came in

Try putting progress instead of answer   הודעה מקורית   מאת: Tony Mountifield נשלח: יום רביעי, 25 בנובמבר 2015 08:14 אל: asterisk-users@lists.digium.com השב ל: Asterisk Users Mailing List - Non-Commercial Discussion נושא: [asterisk-users] Dialing a call back out on same SIP trunk as it came in

[asterisk-users] תשובה: Update new IP address (move temporarily) for INVITE

Use redirect

[asterisk-users] תשובה: Single SIP User on multiple location

Using pjsip you can have multiple endpoints for each extension   הודעה מקורית   מאת: A J Stiles נשלח: יום רביעי, 2 בספטמבר 2015 13:10 אל: asterisk-users@lists.digium.com השב ל: Asterisk Users Mailing List - Non-Commercial Discussion נושא: Re: [asterisk-users] Single SIP User on multiple location

[asterisk-users] תשובה: Asterisk how to setup alarm too many outgoing calls from same user

You could use the group functionCreate the group by extension and check how many calls are in the groupIf it's more than you allow then have it send a email

[asterisk-users] תשובה: Branch based on call volume

Look at the group function

[asterisk-users] תשובה: תשובה: Missed call

It looks like you are dialing a external # then that won't work   הודעה מקורית   מאת: Luca Bertoncello נשלח: יום שישי, 5 ביוני 2015 19:02 אל: asterisk-users@lists.digium.com השב ל: Asterisk Users Mailing List - Non-Commercial Discussion נושא: Re: [asterisk-users] תשובה: Missed call Israel

[asterisk-users] תשובה: תשובה: Accessing an account from more than one phone

from more than one phone Zitat von Israel Gottlieb isr...@gmail.com: Shalom, Israel! Using chan_sip you need to create another ‎user aand then dial both Using pjsip you can connect 2 devices Thank you. Unfortunately it seems that I don't have pjsip available as package on the OpenWRT where I

Re: [asterisk-users] תשובה: Missed call

At the end of the Command you could use options one of them is the c (not apital) which sends a cancel event to the phone http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial On Fri, Jun 5, 2015 at 9:53 AM, Luca Bertoncello lucab...@lucabert.de wrote: Zitat von Israel Gottlieb isr...@gmail.com

[asterisk-users] תשובה: Missed call

If you the c option in the dial command it will send answered else where sip message to the phone and most ip phones understand thatThe cell will always display a missed call

[asterisk-users] תשובה: Accessing an account from more than one phone

Using chan_sip you need to create another ‎user aand then dial both Using pjsip you can connect 2 devices   הודעה מקורית   מאת: Luca Bertoncello נשלח: יום שישי, 5 ביוני 2015 09:24 אל: ML, Asterisk users השב ל: Asterisk Users Mailing List - Non-Commercial Discussion נושא: [asterisk-users]

[asterisk-users] תשובה: Forward loop protection...

We could probably parse the rdnis field to see if it that hop is on the list

[asterisk-users] תשובה: Seeking advice about ISDN BRI Cards

  הודעה מקורית   מאת: jg נשלח: יום חמישי, 28 במאי 2015 12:18 אל: Asterisk Users Mailing List - Non-Commercial Discussion השב ל: Asterisk Users Mailing List - Non-Commercial Discussion נושא: Re: [asterisk-users] Seeking advice about ISDN BRI Cards Thank you all for valuable input, another

Re: [asterisk-users] odbc connection timeout varable

thanks for the reply On Wed, Nov 12, 2014 at 8:03 PM, Matthew Jordan mjor...@digium.com wrote: On Tue, Nov 11, 2014 at 1:43 PM, Israel Gottlieb isr...@gmail.com wrote: well it should but this morning my database hosted at a remote location was down due to conditions at the remote site

[asterisk-users] odbc connection timeout varable

Hi all Does anyone know of a variable that i could check to see if the reason func odbc didnt return results was because of a timeout error so i could play a audio file about that thanks -- _ -- Bandwidth and Colocation Provided

Re: [asterisk-users] odbc connection timeout varable

On Tuesday, November 11, 2014, jg webaccounts...@jgoettgens.de wrote: Why are you concerned? ODBC reconnects automatically if necessary. jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to

Re: [asterisk-users] odbc connection timeout varable

right but that is the problem and i was wondering if there is way for asterisk to set a variable when that happens just like curl On Tue, Nov 11, 2014 at 5:37 PM, jg webaccounts...@jgoettgens.de wrote: Unless of course the database server is not running at all for some reason. But that's not

Re: [asterisk-users] odbc connection timeout varable

well it should but this morning my database hosted at a remote location was down due to conditions at the remote site the question isnt if it should happen or not the questions is there a way for me to know that the odbc query retruned empty because of a connection timeout? in curl i could get

Re: [asterisk-users] Loud Ringers and paging systems...

if you use a papt2 or so spa2101 then you could have alert info set to different lengths or styles of ringers i use that in a dorm with phones and have the phones ring short rings at night so it wont wake up the students On Tue, Aug 5, 2014 at 10:24 PM, Kevin Larsen

Re: [asterisk-users] Get last dialed number in a context?

you could save the info in astdb for the last call per extension and then pull it from there On Tue, Jun 3, 2014 at 12:31 PM, Stefan Gofferje li...@home.gofferje.net wrote: Hi, I would like to implement an auto-redial function in a context. The idea is about like this: Dial a number Hear

[asterisk-users] error cant write to function ODBC_DEVICES

Hi all asterisk 1.8.23 I have odbc all setup to mysql but cant figure out why the dialplan wont write to the odbc function fubc_odbc.conf [DEVICES] dsn=device-conn;dsn in res_odbc not odbc.ini readsql=SELECT call.callNum, call.city, devices.callId, devices.id FROM call INNER

Re: [asterisk-users] Queue callers with Callback option without lose their place

http://www.voip-info.org/wiki/view/Asterisk+Queue+Callback On Fri, Jun 1, 2012 at 1:45 PM, Satish Barot satish4aster...@gmail.comwrote: I believe you want your caller to request for a callback while he/she waits in a queue and when your agents are free, you want to call him back and place

Re: [asterisk-users] concurrent channels limit

are you by chance using the a2billing script? On Mon, Apr 2, 2012 at 5:43 PM, Syco syco...@gmail.com wrote: No, I don't do transcoding, I've disabled all the codec except for the g729. But in my last test I've found out what is the problem (not yet how to solve it) I make all my calls

Re: [asterisk-users] Streaming musiconhold via mpg123

that bug is running since the start of 1.8 and has been fixed in 1.8.9 https://issues.asterisk.org/jira/browse/ASTERISK-17474 i know it says that after the first time asterisks starts it works but thats true only if the moh was loaded before the timing its a long story but the fix is finally in

Re: [asterisk-users] Speech recognition in asterisk using google voice API

wow i just tried in hebrew and i'll say just 1 word WOW On Wed, Jan 4, 2012 at 9:48 PM, sean darcy seandar...@gmail.com wrote: On 1/4/2012 2:26 PM, Lefteris Zafiris wrote: Works beautifully. Amazing job Lefteris. Thanks. The best result I got in probability was 0.9725632 by saying, hello.

Re: [asterisk-users] Not able to play wav files in asterisk

On Mon, Dec 26, 2011 at 9:03 AM, Steve Edwards asterisk@sedwards.comwrote: On Mon, 26 Dec 2011, isr...@gmail.com wrote: Rename the wav to ulaw Miss_audio.ulaw Very bad advice. that might be but if you take a pcm ulaw encoded file and name it .wav asterisk will throw that error I

Re: [asterisk-users] using variables in the shell function

On Wed, Sep 14, 2011 at 5:27 AM, Dale Noll dn...@wi.rr.com wrote: On 09/13/2011 07:49 PM, Israel Gottlieb wrote: is it possible to pas variables to the shell function Set(recordingavail=${SHELL(ls /var/lib/asterisk/sounds/**custom/${TOPMENU})}) im trying to see if a file is available

Re: [asterisk-users] using variables in the shell function

On Wed, Sep 14, 2011 at 4:08 AM, Steve Edwards asterisk@sedwards.comwrote: On Wed, 14 Sep 2011, Israel Gottlieb wrote: is it possible to pas variables to the shell function Set(recordingavail=${SHELL(ls /var/lib/asterisk/sounds/** custom/${TOPMENU})}) im trying to see if a file

[asterisk-users] using variables in the shell function

is it possible to pas variables to the shell function Set(recordingavail=${SHELL(ls /var/lib/asterisk/sounds/custom/${TOPMENU})}) im trying to see if a file is available before playing the file or does anybody have a different idea but not using agi asterisk 1.6.2.20 thanks --

Re: [asterisk-users] asterisk curl and utf8 problems

On Thu, Sep 8, 2011 at 1:52 AM, Israel Gottlieb isr...@gmail.com wrote: On Thu, Sep 8, 2011 at 1:47 AM, Israel Gottlieb isr...@gmail.com wrote: Hi all i have a very weird problem with curl and utf8 characters i'm trying to do a cnam lookup from a web-service with curl if the returned

[asterisk-users] asterisk curl and utf8 problems

Hi all i have a very weird problem with curl and utf8 characters i'm trying to do a cnam lookup from a web-service with curl if the returned info is English or digits then the callerid name field gets populated with that but if the returned info is utf8 like Hebrew then the callerid field remains

Re: [asterisk-users] asterisk curl and utf8 problems

On Thu, Sep 8, 2011 at 1:47 AM, Israel Gottlieb isr...@gmail.com wrote: Hi all i have a very weird problem with curl and utf8 characters i'm trying to do a cnam lookup from a web-service with curl if the returned info is English or digits then the callerid name field gets populated

Re: [asterisk-users] Increasing volume ?

Set(VOLUME(TX)=10) is correct but you arent putting it in a context so asterisk doesnt know how to deal with it do this [bigbluebutton] exten = _.,1,Set(VOLUME(TX)=10) exten = _.,1,Set(VOLUME(RX)=10) exten = _.,n,Goto(start-dialplan,s,1) exten = _.,n,Hangup On Thu, Aug 4, 2011 at 4:33 PM,

Re: [asterisk-users] FAX with SIP

On Fri, Jul 22, 2011 at 12:39 AM, Kevin P. Fleming kpflem...@digium.comwrote: On 07/21/2011 04:34 PM, Joaquin Sosa wrote: On Mon, Jul 18, 2011 at 07:58, Steve Daviesdavies...@gmail.com wrote: The magic sauce that you need is T.38 - Asterisk 1.6 supports this to a limited degree, and your

Re: [asterisk-users] FAX with SIP

On Fri, Jul 22, 2011 at 12:50 AM, Kevin P. Fleming kpflem...@digium.comwrote: On 07/21/2011 04:43 PM, Israel Gottlieb wrote: On Fri, Jul 22, 2011 at 12:39 AM, Kevin P. Fleming kpflem...@digium.com mailto:kpflem...@digium.com wrote: On 07/21/2011 04:34 PM, Joaquin Sosa wrote

Re: [asterisk-users] Help: How can I Add my own Word in option packets in from field of SIP From Asterisk??

user-agent could be set in sip.conf On Wed, Jul 20, 2011 at 12:43 PM, Alex Balashov abalas...@evaristesys.comwrote: On 07/20/2011 05:00 AM, Masood Ahmed wrote: Hello All, Is there any one who can help me to change the From field parameters in option packets, I have seen that in option

[asterisk-users] multiple asterisk on 1 machine or other idea for using multiple network connection

Hi all i have a scenario where i have 2 DSL lines (i know its not that reliable but it fits the bill) connected to 1 box and would like my isp to round robin between both dsl (to allow for more capacity - each dsl could get me thru about 16-18 calls and i need about 30 incoming sip gets routed

[asterisk-users] setting sip headers when using call files

Hi Does anybody have a idea how I could set sip headers when using call files? I have to call out a specific trunk so I cant use local as the trunk what i'm trying todo is send out calls as anonymous but at the itsp it should be filed as being called out thru a specific DID and not the main DID

Re: [asterisk-users] setting sip headers when using call files

On Thu, Apr 14, 2011 at 3:51 PM, Israel Gottlieb isr...@gmail.com wrote: Hi Does anybody have a idea how I could set sip headers when using call files? I have to call out a specific trunk so I cant use local as the trunk what i'm trying todo is send out calls as anonymous but at the itsp

[asterisk-users] how to check if the call is using t38 except in the sip packets

How could i check if the call is using t38 except looking at the sip debug? Is there any variable thats set or could be set? thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join

Re: [asterisk-users] spa8000 spa2102 t38 faxing

t38 ? Thanks On Mon, Mar 28, 2011 at 1:10 AM, Larry Moore lmo...@starwon.com.au wrote: On 28/03/2011 5:48 AM, Israel Gottlieb wrote: still no luck i hear it change to t38 but it just doesnt connect Do you have two fax devices at your end, even a fax-modem attached to a computer

Re: [asterisk-users] spa8000 spa2102 t38 faxing

still no luck i hear it change to t38 but it just doesnt connect On Sun, Mar 27, 2011 at 5:26 AM, Larry Moore lmo...@starwon.com.au wrote: Perhaps this will help. I have a SPA8800 which has 4 x FXS 4 x FXO ports. It took me some time to produce a working configuration. In Asterisk I

[asterisk-users] spa8000 t38 faxing

Hi I'm trying to get the spa 8000 used with a fax machine using t38 passthru i have tried with 1.6.2 and 1.8.3 and is still a no go the provider i use is 012 in israel wich supports t38 (i use it with ffa) could anybody give me a clue how to get this working if it should t38pt is set to yes in

Re: [asterisk-users] extend the timout on ringing for pri or sip

seconds. On Wed, Feb 23, 2011 at 3:17 PM, Israel Gottlieb isr...@gmail.com wrote: Hi Does anyone know how i could extend the timer for the ringing time on a pri or sip trunk ? Today the call gets a cancel request after a minute if not answerd yet is it on asterisk or is a provider side

[asterisk-users] extend the timout on ringing for pri or sip

Hi Does anyone know how i could extend the timer for the ringing time on a pri or sip trunk ? Today the call gets a cancel request after a minute if not answerd yet is it on asterisk or is a provider side setting? -- _ --

[Asterisk-Users] How do I add a list of cidnames to the asterisk database in one shot ?

but I cant seem to find it). Thanks, Israel Gottlieb ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE