search for: morreni

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

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

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 am having a strange problem with attempting to get voicemail-to-mysql to work. The biggest problem is that I am not able to store voicemail into database. So, I followed the instructor found on the web: Updated the /usr/src/asterisk/apps/Makefile to have USE_MYSQL_VM_INTERFACE=1 and recompiled asterisk, with make clean; make; make install (By the way, is it necessary to update the Makefile

Dear all, I am able to send and receive fax with Asterisk + iaxmodel + hylafax. What I want to be able to do is to 1. Stored the received fax in mysql 2. Send an email notification to he user corresponding to the incoming phone number 3. Send a SMS notification to the user's mobile phone In the hylafax setup, it seems like it can only send email to one destination email address. Is there

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 Asterisk-User friends, After realtime queues are defined, how does it work with the agents? There seems to be no db table for agents. If I can't define agents for the realtime queues in the db, how can agent login/logoff be done? Thanks alot for your help. Thanks, Mark -------------- next part -------------- An HTML attachment was scrubbed... URL:

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 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

CLI>realtime mysql status Connected to asterisk at 127.0.0.1, port 3306 with username askuser for 1 minutes, 34 seconds. Thank you very much for your kind attentino. You help is greatly appreciated. Thanks, Mark ------=_NextPart_000_05AD_01C88E72.41751FF0 Content-Type: text/html; charset="us-ascii" Content-Transfer-Encoding: quoted-printable <!DOCTYPE HTML PUBLIC

Hi, I am having problem with my Asterix server. It does not hand up after play the voicemail. The scenario is this: 1. I make a call to Asterisk's PSTN number; 2. After recording, I hang up and make the same call again. The first call would go through nicely with the voicemail recording, but the second call will hit a message saying "the other party is busy". The only way to fix

Dear friends, I am wondering if there is any efficient way of extract the country code, area code, and local code into 3 different variables from one DNID that can look like 001630233-4333 or 0086213345333? International code can be 011, or 00. National code can be 0 or 1 Country code can have 2 or 3 digits Area code can have 2 or 3 digits Local num can be 7-10 digits Is there anyway to break