search for: markmorreny

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, 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 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 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 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 friends, I have a question about Asterisk usage. Is there anyway to configure Asterisk in such a way that once a call is forwarded out 1: to land line; or 2: to another SIP server, then the channel will be released and free up? The senarior is let's say I have an incoming ISDN line with limited channels, when all my lines are busy, I can leave one channel always open to redirect calls.

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 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

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, Sorry to resend the same question. This mail is just to add a few bits of details: When I tried to join the "support" queue, I get L RealTime: Retrieve SQL: SELECT * FROM queue_member_table WHERE interface LIKE '%' AND queue_name = 'Support' ORDER BY interface [Mar 29 10:01:52] WARNING[6203]: app_queue.c:3939 queue_exec: Unable to join queue 'Support' In

Dear friends, I am trying to configure Asterisk so that it play differnt set of voicemail greets for differnt extensions. I put my customized .wav files under the extension, but it still does not work. Asterisk still plays the default voice file. debian:/var/spool/asterisk/voicemail/default/2000# ls -al total 116 drwxr-xr-x 6 root root 4096 2008-04-01 06:03 . drwxr-xr-x 7 root root 4096

Dear all, I noticed a very strange problem. When I tried using VoiceMailMain to record my unavailable message, the file does not get created even though I can find the corresponding mssage from asterisk: -- <SIP/2001-b6307d78> Playing 'beep' (language 'en') -- x=0, open writing: /var/spool/asterisk/voicemail/default/2000/unavail.tmp format: wav, 0x82828c8 --

Dear all, I am having a very strange problem with VoicemailMain. When using this application to record unavail, greet, and busy, I an see the corresponding file gets created in the <..>/default/<SIP #> directory. When pressing "1" to confirm the recorded message, the *.wav file gets deleted from the file system. How can this happen? I can't figure out why. Is there

Hi, We are running SPA 3102 in multiple places and the one that we have problem with is with China telecom. Does anyone know the correct disconnect tone setting for China? Thanks in advance. Mark -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080604/23458301/attachment.htm

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

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