similar to: Asterisk not hanging up after voicemail

Hi, Is it possible to connect Asterisk with a mobile base station to handle call switching? What kind of protocol will I need to use to convert to sip? Any pointer or info will be greatly appreciated. Best Regards, Mark -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081119/e74ef6b1/attachment.htm

Hi, I have a Linksys SPA 3102 using as ATA. The call routing is : Phone -> PSTN -> SPA 3102 -> SIP Proxy -> Asterisk The problem I am having is that when the phone hangs up, SPA 3102 can't detect it and relay the CANCEL message. Is this problem with my SPA 3102 config or it just works like that by default? Thanks in advance for your help. Regards, Mark -------------- next

Hi I am working on deploying voip for my company and would like to seek some advice on the number of E1 lines we need to rent. Our telco told us that there can be at most 30 concurrent channels on an E1 line. Typically, what is the maximum number of DIDs that we can allocate to that E1 line before users get frequent "all lines are busy"? We are running a support center with mostly

Dear all, I am working on customizing hylafax's faxrcvd script into PHP. Does anyone has any sample or guideline that can share with me to give me a quick start? Two questions I have are: 1. How to simulate the receival of fax without actually sending one? 2. Where can I find the log that is "echo" from faxrcvd? 3. How to I config Hylafax so that it uses my PHP script instead

Hi all, I am using Digium PCI board to receive PSTN call through regular phone line. It is no problem for me to receive calls, but I am not able to obtain the Caller ID if the calls are from the phone line. exten => s,1,Answer() exten => s, n, Verbose(1|incoming number is ${CHANNEL} calling to ${EXTEN} routing to ${phonenum} ) exten => s,n, Verbose(1|callid is ${CALLID(num)}) exten

Dear friends, I am having problem with running a sample php and I can't figure out why. I can run the sample.php using CLI but when I run it inside the dialplan it does not work. Can someone please suggest the config problem that I may have made? dommy:/var/lib/asterisk/agi-bin# php sample.php #!/usr/bin/php5 -q VERBOSE "Here we go!" 2 VERBOSE "Call from - Calling

Hi, I have two asterisks that are connected to each other via a back-to-back E1 link using a pair of sangoma cards. With the following scenario: SIP-PHONE <-> Asterisk <-> E1 <-> Asterisk <-> SIP-PHONE, the sound quality degrades significantly. I can't understand why as the amound of packet lost should be very minimum. Does anyone know why? Does it have anything

Hi, I am simulating the sending of fax using sendfax through voip to reach an Asteria server via ZAP/1 ( PSTN phone line ) which then route call to a fax machine at ZAP/2. It seems like I am not able to establish any handshake with the physical fax machine using the sendfax program. Does anyone know why that happens and how to fix it? The scenario will be deployed in remote location in the

Dear all, I am still not able to store voicemail into mysql and I am hoping someone can help me out. Here is my voicemail.cof: [general] format = wav attach = yes dbuser=ast dbpass=sqlpass dbhost=localhost dbname=asterisk odbcstorage=asterisk odbctable=voicemessages [default] ; Syntax for new entries looks like this: ; MailboxNumber => password,name,e-mail,pager,options ; (usually, the

Dear friends, Asterisk's voicemail functions work fine for me, but I am having difficulty storing the voice messages inside mysql. My real-time CDR recording works so I assume the odbc connection is fine. The voicemail.conf I have is : [general] format = wav attach = yes dbuser=root dbpass=sqlpass dbhost=localhost dbname=asterisk odbcstorage=asterisk odbctable=voicemessages Asterisk shows

Hi, I am looking for a very low cost way of receiving and sending T38 fax reliably. Is there any possible solution using Asterisk as the PSTN SIP gateay and Digium E1/T1 card? Is there other open source package that can help to accomplish this purpose? Regards, Mark -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, This is not a repeated post as I am just adding more information for my previous post. Asterisk version 1.4.18 TDM card: Digium TDM411B Zaptel version 1.4.9.2 Line: PSTN line I tried to obtain the dialed number using $DNID and $CDR(DST) . All of these variable returns 's' I also tried exten => _3345335,n,Noop(this is ok) where 3345335 is my number but it does not go there.

Hi all, I am using Digium PCI board to receive PSTN call through regular phone line. It is no problem for me to receive calls, but I am not able to capture the destination number through the ZAP channel exten => s, n, Verbose(1|destination to ${EXTEN} ) ${EXTEN} returns 's' instead of the actual destination number. Since I have multiple phone numbers, I want to be able to route

Hi, I have read about SS7 recently and learnt that it is a signalling protocol used in PSTN for call management, setup, etc. The thing that I don't understand is how SS7 plays a role in VOIP. When I make calls between landline and Asterisk via PSTN, I don't need to do anything with SS7. Is it because the SS7 signalling is already done by Asterisk already? From the prespective of

> Le 20 oct. 2019 ? 22:24, Mauricio Tavares via dovecot <dovecot at dovecot.org> a ?crit : > > On Sun, Oct 20, 2019 at 10:43 AM Rajesh Bansal via dovecot > <dovecot at dovecot.org> wrote: >> >> Hi Team, >> >> >> >> I need to develop Visual VoiceMail solution. In this solution I need a IMAP4 server, from which I can get a hit for each

Hi Team, I need to develop Visual VoiceMail solution. In this solution I need a IMAP4 server, from which I can get a hit for each mail retrieval. Can anyone help me if dovecot can be used for this purpose. BR, Rajesh Bansal -------------- next part -------------- An HTML attachment was scrubbed... URL:

Folks, We have several people using SIP softphones in the office. When they leave for the day, they power down their workstations, causing their registration with Asterix to quickly timeout. Here's the entry for one such extension in extensions.conf: exten => 8102,1,Dial(SIP/someone,20) exten => 8102,2,Voicemail(u8102) exten => 8102,3,Hangup exten => 8102,102,Voicemail(b8102)

If you use an Iphone (and your mobile proivder supports it) then you are using it ;-) just my 2 cents Am 21.10.2019 um 12:26 schrieb Stephan Bosch via dovecot: > > > Op 20-10-2019 om 22:33 schreef Jean-Daniel via dovecot: >> >> >>> Le 20 oct. 2019 ? 22:24, Mauricio Tavares via dovecot >>> <dovecot at dovecot.org <mailto:dovecot at

Hi Timo, I have another problem, this time with user shared folders: User "markus" shared the folder "ForTest" to test: SETACL "INBOX/ForTest" test akxeilrwts dovecot-acl and shared-mailboxes.db have been successfully updated. As user "test" the folder (and "#User" namespace) is not visible. When I configure the shared namespace with

I can either get the TE100P working or the 3 X100P clones working, but never both. I have the TE100P connected to a channel bank, and X100P clones to lines from the phone company. This is my zaptel.conf span=1,1,0,d4,ami fxsks=1-24 loadzone=us fxols=25-27 loadzone=us I then do [root@asterix root]# modprobe zaptel [root@asterix root]# modprobe wcte11xp ZT_CHANCONFIG failed on channel