similar to: Calls dropping

Hi All, I am relatively new to Asterisk. I have CDR enabled and successfully writing to MS SQL server. In my cdr table I am setting the userfield value with a line in my dialplan. If a call is placed to an invalid number (e.g. 12125551212), I see a cdr record created, however, my userfield value never gets set since the call never made it into the context of my dialplan. I am using AMI with the

Hi, I am attempting to implement Answering Machine Detect and have also played with using BackgroundDetect instead. Does anyone recommend one over the other? Here is the code I am using for the BackgroundDetect method (from voip-info.org). Thanks. [detect] exten => s,1,Set(MACHINE=0) exten => s,2,Answer exten => s,3,BackgroundDetect(silence/5, 1000, 50) exten =>

Hi, I just deployed new Aastra 9480i phones and when I attempt enter digits on other systems, like host pin in a GoToMeeting, the servers on the other end do not get my entries. I am assuming this is a DTMF issue but do not see anything in this phones config other than turning on the display of the digits. I have the DTMF method set to "SIP INFO". I am using AsteriskNow w/FreePBX.

Hi all, I am trying to make decisions in my dialplan based on {DIALSTATUS}. I am creating calls using AMI (rawman with parameters via URL) with action:Originate. I am using SIP and an outside voip provider for the calls. If I define the number to call in the Channel parameter (e.g. SIP/15555555555 at myvoipprovider, the call gets placed before entering the context that I defined. I understand

Hi, As of v 1.6.1.1, can anyone tell me what the current possible values for DIALSTATUS could be? I found http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS but believe it is outdated since there is no FAIL or FAILED in this list. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL:

I upgraded our Asterisk 13 LTS to Asterisk 16 Certified 16.8 yesterday and converted form SIP to PJSIP using the python script as a start and then mofiying from there.  I ran into an issue when testing that incoming calls from MagicJack would go silent after about 10 seconds.  This happened while in the automated attendant area.  This problem did not occur with Asterisk 13 LTS.  I reverted PJSIP

Hi I was interested to see how Mongrel uses Lex/Yacc to parse the HTTP requests using a Regal generated parser. I downloaded the source but do not see the lex and yacc files...

Does anybody have T.38 faxing working with Flowroute? I am running Asterisk 1.6.2.7-rc3 with spandsp 0.0.6-pre17. I can successfully receive faxes over ulaw. I enabled T.38 with t38pt_udptl=yes in sip.conf. When I receive a fax it tries to negotiate T.38 and Flowroute sends back a Bad Request response saying I have a SIP syntax error. Flowroute support is recommending that I try again after

Asterisk 1.4.32 dahdi-2.3.0.1 Centos 5.5 Digium TE420 CAC channel bank (2) Cisco RVS4000 router Cox 50 Mbps/ 5 Mbps cable modem Flowroute provider codac G-711 90 % CPU idle callwaiting=no When there are 10-15 or more calls up the farend hears a callwaiting like beep every 3 to 6 sec. the duration of this "beep" is very short and would be no problem if it didn?t happen every few

Hi, please forgive me for this (hopefully) simple question. I cannot seem to find an answer or solution while searching around. I want to be able to call in to my server using my cell phone and be able to set call forwarding for my extension and enter a phone number and also be able to call in to that extension and disable the call forwarding. I see I can do this through the ARI web interface

I have an Asterisk 1.4.2 system that has been installed for about 3 months now in our home. We converted all of our phones to SIP phones, and use two different trunk providers (BroadVoice for incoming & FlowRoute for outgoing). Most of the time its working flawlessly. But about 1/3rd of the calls that come into us complain of an echo and what is best described as latency issues. Its

Hi, I am attempting to make about ten calls simultaneously and intermittently get 'SIP/voipprovider is circuit-busy' followed by 'everyone is busy/congested at this time" I am not sure if this is related to my bandwidth to my voip provider, a configuration issue or something else. Has anyone seen this before and have any suggestions. Thanks in advance. --------------

Hello List, Since I'm looking for a new VoIP provider for US origination/termination, I will very appreciate if you can chare your experience with Flowroute, Vitelity and Voip.ms Thanks in advance! Elder D. Arohuanca dCAP 1497 Lima - Peru -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I'm having a problem trying to do this, and it used to work with Asterisk 1.4. Since Asterisk 1.6 series I have not been able to place more than one call to a DID. I get this message: Skipping dialing interface 'SIP/16034817531 at flowroute' again since it has already been dialed I'm trying to do this because many of my clients have roll-over lines and they need to

Hi, I've been getting complaints lately that callers to my IVR are pressing a digit once but the system is responding as if they pressed it twice (once for each of two consecutive menus). I'm using an AGI script and logging all DTMF entries - and to the script, at least, it looks like the digit is being pressed twice. The TN being called is a VOIP number (provided by Flowroute) and being

I have a block of DID's that I ported to Vitelity about 7 days ago. The problem is if a POTS caller dials into the system, his dtmf is not heard at READ() or Background() while a prompt is played. After the prompt is finished, then dtmf is heard. I've been working with their support, but it still not resolved. SIP callers are not effected. Yesterday, I purchased a DID from

Can anyone comment on using SMS in conjunction with VoIP service using one of these three VoIP providers: voip.ms, vitelity.com, flowroute.com? Are some SMS services more compatible with Asterisk (i.e. SMS over SIP works perfectly or not)? Is it best to use a different data channel for SMS messages (i.e. SMS via HTTP, SMS via XMPP) instead of Asterisk's built in SMS application

What is the best way for me to setup Fax Capability with VOIP only. I have a Asterisk Server hosted on the internet without a modem. I'm using Flowroute, which is working awesome, for VOIP calls. I only have a SIP Phone at home and two Printer/Scanner/Fax Printers. I'm not sure which Fax Addons or Extensions I should use for Asterisk. I'd like it to Self Detect on any line. I

We currently use asterisk 1.4.x with two Zaptel cards connected to POTS lines. So we make "outbound" calls from their softphones (using ulaw format), which go over a dedicated DSL line to the asterisk server in our office, which then converts the calls to POTS. This all works fine, assuming there aren't any unusual problems. It sounds as good as POTS on both ends. However, we

Hi, My Dial() is set to the following, but always stops about 30 seconds into the call even when I set it to try for 60 seconds. exten => dialnumber,1,Dial(${DialInfo},60) I am running on 1.6.1-r199820. Is there some other setting that is overriding mine? Or an issue with this release? Thanks for the help. JR -------------- next part -------------- An HTML attachment was