Re: [asterisk-users] withheld caller id

On 2018-04-10 10:19, Atux Atux wrote: exten => _9X.,1,Dial(Dongle/dongle800/#31#${EXTEN:1},120,KT) exten => _9X.,n,Hangup(${HANGUPCAUSE}) What am i doing wrong in asterisk? unless i'm missing something your config looks OK. Do you have any logs / debugs of what number is actually being

Re: [asterisk-users] withheld caller id

On 2018-04-10 08:46, Atux Atux wrote: 9+#31#+destination_number. Unfortunately, zoiper did stop on 9#31# and it dialled one of my recent numbers. The same result happened with haven't used zoiper at all, so can't comment on its features of parsing numbers. I'd recommend 'hiding' this

Re: [asterisk-users] bash: asterisk: command not found

On 2016-12-07 09:13, Steve Howes wrote: On 07/12/16 04:56, christopher kamutumwa wrote: Ive installed asterisk 14.2 on centos 6.8 but i am not able to start it below is what am executing and those are the errors anything am doing wrong? It doesn't look like it is installed to me... Check the

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

On 2016-08-04 09:12, Nabeel wrote: The problem is that the 'mailbox' prompt allows a way for accessing any other mailbox, which is not necessary in my case. If anyone knows a way to remove this 'mailbox' prompt, please let me know. Not having followed the thread from the beginning I must

Re: [asterisk-users] "Follow me" with Asterisk that detects cellphone voicemail and similar announcements

Theoretically, I could use the dial function to call one number, then wait a few seconds and then dial another number. In practice, this won’t work because as soon as a call is answered by the mobile carrier’s voicemail the caller would be connected to that, no other numbers would be called.

Re: [asterisk-users] Call Center

I've done something like this, and it was done by an external perl script. Asterisk played its part through call files to initiate calls and putting them into the right point in the dial plan, playing sounds and accept user iteraction (through dial plan extensions or AGI). Totally doable

[asterisk-users] a2billing

hello, I just installed a2billing, I did all the config, at least I guess .. but I still can not integrate sip accounts that I had created (with sip.conf ) in a2billing to apply their billing .. could someone tell me how to make the junction between asterisk and a2billing?? I also noticed that the

Re: [asterisk-users] a2billing

I confess that I had already generated sip accounts but I just delete them. it remains in the dialplan a2billing and RateCard I created .. Please can you tell me how to complete the setup? 2012/6/22 Mikhail Lischuk mlisc...@itx.com.ua ** Gorguez Ka писал 22.06.2012 15:31: hello, I just

[asterisk-users] urgent

Hello, I would do the billing on Asterisk with A2Billing, but at the configuration of the mysql database when trying to display the table I created with this command: mysql-u root-p mya2billing with mysql show tablename since there is not even able to show tables I then return - And nothing else

Re: [asterisk-users] urgent

] *Im Auftrag von *Gorguez Ka *Gesendet:* Mittwoch, 20. Juni 2012 14:52 *An:* asterisk-users@lists.digium.com *Betreff:* [asterisk-users] urgent ** ** Hello, I would do the billing on Asterisk with A2Billing, but at the configuration of the mysql database when trying to display the table

[asterisk-users] Astmanproxy Yahoo Group

Hi All, Is the astmanproxy yahoo group (http://tech.groups.yahoo.com/group/asterisk-astmanproxy/) still working? It seems to me the most recent posts are 2006's. I have sent a message but didn't receive any feedback and the post was not listed. Is the project still being maintained? Is there

[Asterisk-Users] Asterisk Keep Crashing need Help please

Hi All, My Asterisk Server crash at least once a day. The full and message log files did not show any errors or indication on why the asterisk service crashed. After looking at the core dump file, I am still unable to identify the problem. Can someone please give me some guideance on what I can

[Asterisk-Users] Listening to agent's conversation while waiting in the queue

Hello, I would like to configure a queue with only one agent. But while waiting the callers in the queue should listen to the agent's conversation instead of a music on hold. Is that possible with Asterisk? Does someone out there ever work on a similar configuration? Thank you for any suggestion

[Asterisk-Users] Hang up as soon as other party picks up call

Hello, I have an Asterisk box with a TE410P connected to a PRI line and agents with X-Lite installed on the same LAN as the Asterisk server. Sometimes, when I make outbound calls it hangs up as soon as other party tries to picks up the call. Does someone ever experienced this situation? On

[Asterisk-Users] Prad or V5.2

Hello, Does someone out there ever heard about PRAD or V5.2. Is there any link with Digium's TE110P? Thanks for any enlightments Lamine ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

[Asterisk-Users] Invalid sip contract

HI, all I receive the message from asterisk cli likes below,I don't know what makes it.Can anybody help it.Thanks!!! May 19 10:14:21 NOTICE[3799]: chan_sip.c:4814 parse_ok_contact: '' isnot a valid SIP contact (missing sip:) trying to use anywayMay 19 10:14:21 WARNING[3799]: chan_sip.c:4847

[Asterisk-Users] Junk at the beginning, Warning, flexibel rate not heavily tested!

Hi,all I am newer to Asterisk.My Asterisk version is the newest CVS-HEAD.now something appears in the console CLI like below these,I don't know what's happen to my Asterisk Server.Could anybody help me? Thanks Junk at the beginningWarning, flexibel rate not heavily tested!Junk at the

Re: [Asterisk-Users] How to decrease Asterisk load

Should I believe that at this time there is no DSP capable cards working with Asterisk? - Original Message - i From: izo [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, May 13, 2005 6:11 AM Subject: Re:

Re: [Asterisk-Users] IPVolution release info....

Thanks for this precision !! Certainly, a good news for Asterisk users community. - Original Message - From: Wiley Siler To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, May 12, 2005 10:16 PM Subject: [Asterisk-Users] IPVolution

[Asterisk-Users] How to decrease Asterisk load

Hi everybody, I would like to decrease the load of my asterisk server. Could someone recommend me a solution? I have thought about a hardware component that would do some tasks as compression/decompression or codec translations but wonder if such a solution exist. Thanks for any suggestion

Re: [Asterisk-Users] How to decrease Asterisk load

Thanks Mike, I am already using rawplayer for music-on-hold. I have been told of IpVolution TDM60 card that has DSP resources ... Does someone out there ever experienced it? Lamine - Original Message - From: Mike Holloway [EMAIL PROTECTED] To: Mamadou Lamine KA [EMAIL PROTECTED]; Asterisk

Re: [Asterisk-Users] Voice Quality

Hello David, Bad voice quality may be caused by many reasons. I suggest you test the two servers separately first. Monitor CPU load during calls in each server and verify if the communication devices used by asterisk (voice boards, network interfaces ... ) don't share interruptions. In iax.conf

[Asterisk-Users] ACD in Asterisk

Hi everybody, I am having a problem while setting up queues in Asterisk. Callers are kept in the queues and told to wait while there are available agents. Even if I use ringall as strategy the call is not always sent to all free agents. Is there a problem with Automatic Call Distribution in

[Asterisk-Users] Failed to authenticate

HI,all! I have setted up an Asterisk server on my linux.My sip.conf and extensions.conf configurations like these: sip.conf [general]port=5060disallow=allallow=alawallow=ulawallow=gsmnat=yes dtmfmode=rfc2833canreinvite=no

[Asterisk-Users] Failed to authenticate

HI,all! I have setted up an Asterisk server on my linux.My sip.conf and extensions.conf configurations like these: sip.conf [general]port=5060disallow=allallow=alawallow=ulawallow=gsmnat=yes dtmfmode=rfc2833canreinvite=no

[Asterisk-Users] ast_expr.y:243 to_integer:Overflow

hi,all! Can anybody tell me what's the matter? thanks!!! Do You Yahoo!? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

[Asterisk-Users] chan_sip.c:7174 handle_request : Failed to authenticate user

Hi,all ! My asterisk's console appear some words : "chan_sip.c:7174 handle_request : Failed to authenticate user "top" sip:1002:@10.0.0.1 tag=169447308" . Can anybody tell me what cause it ? thanks!!! Do You Yahoo!? ___ Asterisk-Users mailing list

[Asterisk-Users] Asterisk Addons compile errors

HI: I have compiled and installed Asterisk 1.0.7 without any problems.I have also installed mysql and DBD::mysql successfuly / When I tried to make asterisk-addons, it showed me the problem like these: [EMAIL PROTECTED] asterisk-addons]# make install ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE

[Asterisk-Users] joinempty=no

Hello, I would like to know if there is a simple way to redirect callers to another extension (may be an IVR) when no agent is logged on. joinempty is set to no in my queue. Thank you for any tip Lamine ___ Asterisk-Users mailing list

[Asterisk-Users] Chanspy and current version of cvs

Hi everybody ! I had patched asterisk to install chanspy weeks ago and everything was ok. With the current version of cvs i am having failures when i try to apply the same patch and the url where i originally downloaded it seems no longer active. Is the patch any longer maintained or has it been

Re: [Asterisk-Users] Call asterisk from perl

You can also use the manager. Take a look at http://www.voip-info.org/wiki-Asterisk+manager+API - Original Message - From: Kevin P. Fleming [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, February 16, 2005 6:12

[Asterisk-Users] TDMO4B, GSM Gateways and CallerID

Hello everybody, I have an Asterisk box with a TDM04B and would like to connect it to a GSM Gateway. Can someone tell me whether i can get the callerid for incoming calls in this case? Thanx Lamine ___ Asterisk-Users mailing list

[Asterisk-Users] Agents question

Hi, Is there a way to know whether a logged agent is in communication or not? I would like the supervisor to select the agent he wants to spy. Thanks for any suggestion Lamine ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

[Asterisk-Users] ChanSpy Usage

Hi, Could someone tell me the significance of arguments in chanspy synopsis: Chanspy([-opts|]chan_name|scan[|scanspec]). What are the possible values for opts,scan and scanspec? Thanx Lamine ___ Asterisk-Users mailing list

Re: [Asterisk-Users] Signaling / Streaming

Hi, With Gnugk, make sure the proxy mode is not enabled if you want voice to pass directly from endpoints. Regards Lamine - Original Message - From: Joao Pereira [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday,

Re: [Asterisk-Users] Signaling / Streaming

to Asterisk makes the diference. If we are using them as proxy, the stream will pass through them, if we dont use proxy, they will be used just for signaling. Joao - Original Message - From: Mamadou Lamine KA [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial

Re: [Asterisk-Users] Monitoring

What version of sox do you use? Lamine - Original Message - From: Robert Spielmann [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, January 07, 2005 2:40 PM Subject: [Asterisk-Users] Monitoring Hi, I have some trouble with the Monitor() application. I start and stop

Re: [Asterisk-Users] TDM400 problem

As you have a TDM400 you should load wcfxs module ( modprobe wcfxs or modprobe wctdm for new versions) regardless to modules you have installed on your board. You should also check the signalling specified in zaptel.conf according to your modules and the order they are placed on your TDM400. Best

Re: [Asterisk-Users] where I can find some learning book about asterisk?

Hello, Take a look at http://www.signate.com You can also find various documentation resources at http://www.voip-info.org/tiki-index.php?page=Asterisk Regards Lamine - Original Message - From: FCG ZHAO Zigang [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday,

Re: [Asterisk-Users] asterisk at large

Hello *'s, First Of all Marry Christmas, I want to setup asterisk at large means my main asterisk server placed in my office(in Pakistan), and some offices outside Pakistan and i want to connect these locations to my main * server (in Pakistan) on remote locations i'll used asterisk can i

[Asterisk-Users] Problem with sox

Hello, I have installed sox-12.17.6 from sources with 2.4.27 kernel but i can't mix audio files. (For both wav and gsm formats) For example, when i try soxmix filename-in.wav filename-out.wav filename.wav. Everything seems ok. There no error messages. But the output file (filename.wav) is empty,

[Asterisk-Users]Re: Problem with sox

Thanks Dawson, 12.17.5 version works fine but i have got to mix files in gsm format first before converting it to wav by using sox. But soxmix doesn't work directly with wav files. Altus, try using groups in zapata.conf. This can be done by adding group=1 callgroup=1 pickupgroup=1 before

Re:[Asterisk-Users] Setting jitterbuffer in with iax

Hello everybody; I would like to know the parameters on which depend jitterbuffer in iax.conf. Is there some kind of formula to set the correct values? Thanks in advance for any help Lamine I'd say that the numbers in the iax.conf.sample are a good balance. You'll also find quite a

[Asterisk-Users] Setting jitterbuffer in with iax

Hello everybody; I would like to know the parameters on which depend jitterbuffer in iax.conf. Is there some kind of formula to set the correct values? Thanks in advance for any help Lamine ___ Asterisk-Users mailing list [EMAIL PROTECTED]

[Asterisk-Users] VICIDIAL and IAX

Hello everybody, I would like to know if there is a support of IAX in vicidial. I want to make predictive dialing use vicidial using IAX soft phones. Thanks in advance Lamine ___ Asterisk-Users mailing list [EMAIL PROTECTED]

[Asterisk-Users] ZapBarge and SIP Channels

Hello everybody, Is there any alternative to Asterisk ZapBarge command for SIP and IAX channels? Thanks Lamine

[Asterisk-Users] fax detection and X100P

i have successfully updated my cvs pull of zaptel but for asterisk when i type "make clean"i have the folowing error: Makefile:73: *** missing separator. Arrêt ( Arrêt means stop) Lamine

[Asterisk-Users] fax detection and X100P

Hi everybody I am having problem detecting fax with my X100P. I have RedHat 8 as OS and an X100P and a TDM400P. The X100P being plugged into PSTN. I have successfully installed tiff-v3.5.7 and spandsp-0.0.1 and also patched Asterisk wthout problem. Here is my zapata.conf file context=cda

[Asterisk-Users] Detecting Events in queues

Hi everybody, I would like to know how can I detect events in queues. For example when an operator answers a call which was in a queue I would like to some informations related to the caller (quote from the database) to the operator. I saw these events in queuelog.conf but i don't know how to

Re: [Asterisk-Dev] SPAM MESSAGE - [Asterisk-Users] warning message (sound card) - when I run asterisk!!!

Hi Neo, Your sound card is not well configured. Try to find the right driver and load the correct module for it. ALSA (www.alsa-project.org/) may help for this. Hope this can help Lamine - Original Message - From: Neo Jia To: [EMAIL PROTECTED] ; [EMAIL PROTECTED]

[Asterisk-Users] Call recording between SIP phones

Hi everybody, I have been searching around for days on how to record calls between SIP phones.Could someone tell me whether it is possible? The Record command doesn't seem to work during a call. Thanks Lamine ___ Asterisk-Users mailing list [EMAIL

[Asterisk-Users] Dial and MeetMe on the same channel

Hello everybody, I would like to know whether it is possible to run Dial and MeetMe commands simultaneoously on the same channel. I am using a C AGI as below but it seems to me that only the first command that is called in the agi is executed. ... // Préparation de la commande pour

[Asterisk-Users] Newbie Start Question

help will be highly appreciated. Thanks in advance Mamadou Lamine KA