search for: backeberg

I'm considering a situation where I buy about twenty ATA devices. I've played with the Linksys / Cisco PAP2T, and got that working fine with some inbound and outbound faxing. The web GUI was okay. I'm seeing prices around $45 to $50 for this thing. It comes with two FXS ports, but I only need one FXS. I've seen the Grandstream Handytone 286 online. It looks promising as an

I'm having annoying errors trying to get configure working. tar xvzf /usr/local/src/asterisk-1.8.6.0.tar.gz cd asterisk-1.8.6.0 ./configure I get complaints related to pwlib / ptlib... checking for openr2_chan_new in -lopenr2... no checking /root/pwlib/include/ptlib.h usability... no checking /root/pwlib/include/ptlib.h presence... no checking for /root/pwlib/include/ptlib.h... no checking

Hello List: I'm working on a funny scenario, where I'm bouncing calls from a Cisco call center into asterisk. Cisco call center has some logic that if a customer calls in, an agent is logged into a given extension... if Cisco sends a customer call to that extension, and there is a ring with no answer after a preset amount of time, Cisco concludes the agent is unavailable, kicks the agent

I've just upgraded to 1.6.2.6 on one of my test systems. I started out happy, with some improvements in transfers to Local() channels from a SIP channel, and much nicer verbose fax handling. However, something is really weird when I need to do System() calls. It was really, really weird. This was also affecting AGI, when I needed to read system variables from asterisk into an AGI Perl script.

Hello all, I've played with background and play sounds apps and googled around and asked the list before to no avail. Does anyone know of a way to have tones played during the call progress stage of the call? We (especially on some international circuits) get up to 5 seconds of silence before the phone starts ringing or is busy. I don't want to force "R" on the Dial app as

Hi, I have encountered a DTMF issue. My scenario: Access carrier-----sip----> Asterisk-1.4.25.1-----sip---->CiscoGW-----ISDN----->TDM Switch the access carrier sends to Asterisk out of band (rfc2833) dtmf, Asterisk forwards it with SIP INFO method to Cisco gateway, but on TDM switch every digit is duplicated. Is it possible that the carrier sends inband along with rfc2833? Kind

http://saunderslog.com/2009/03/03/voxeo-launches-tropocom-mashup-platfor m/ I'll pay anyone a $20 bounty for someone to replicate the USA Asterisk Weather App on Tropo. Would like to see how quickly this is implemented. Regards, Dean Collins Cognation Inc dean at cognation.net <mailto:dean at cognation.net> +1-212-203-4357 New York +61-2-9016-5642 (Sydney

Hi everyone, Using Asterisk 1.6x here with a TDM PRI. I have to run a campaign for about 5000 numbers and then put the call to agents right away and pull up the CRM based on the number dialed. So, I am going to be doing some PHP+Ajax work. I am familiar with spool files but I don't like the fact that I can't read the status of the call in real-time. However, I know that it's the

...age----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steve Underwood Sent: Friday, March 13, 2009 10:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Ast/Hyla/IAX Scalability? David Backeberg wrote: > On Fri, Mar 13, 2009 at 2:30 PM, Marshall Henderson > <marshallmch at gmail.com> wrote: > >> On Fri, Mar 13, 2009 at 11:07 AM, David Backeberg <dbackeberg at gmail.com> wrote: >> >>> Again, you'll find people arguing that their voip sol...

I did an upgrade to the SVN trunk on the 12/9 and when I looked in my mysql table for CDR's today there are no entries since the update. I have rebuilt and re-installed and re-started asterisk still no CDR's flowing to mysql. I did not change any configs. I checked to make sure that the cdr_mysql option was selected under the make menu options. The module shows it is there when I do a

I am testing out inbound faxing using res_fax and res_fax_spandsp.so My system answers the call but then sets there on the ReseiveFax line then comes back with an error that it exceeded the maximum retries. How would I go about debugging this? Below is my very simple dialplan code I am using, and the fax show version gives the following as well. FAX For Asterisk Components:

Hi, I`m occassionally seeing CDR(accountcode)'s value empty at a place in my diaplan where it was filled with some value a few lines before, with nothing else having changed it. It`s giving me headaches (as I rely on it for MySQL queries). Anything I can do? Mick -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello there: I think I have a silly kernel configuration problem. I'm running: * vanilla 2.6.27.10 kernel built from source * dahdi-2.1.0.4 built from source So far so good, dahdi module loads just fine: dahdi: Telephony Interface Registered on major 196 dahdi: Version: 2.1.0.4 when I try to: hal04 dahdi # modprobe dahdi_dummy FATAL: Error inserting dahdi_dummy

Hi all, I've cross-posted this to the -users and -biz groups. Hope that's OK. I have a customer who REALLY needs to be able to send/receive faxes reliably. I could probably get hylafax configured, but I'm not sure how reliable it is. If it is considered reliable, would someone let me know? Otherwise, is there a product/service they can buy that will allow them to fax to/from

Date: Mon, 23 Jun 2008 08:00:08 -0400 From: "David Backeberg" <dbackeberg at gmail.com> Subject: Re: [asterisk-users] Replace music-on-hold on MeetMe with ringing sound To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Message-ID: <3de056a30806230500k7e66185l7...

So historically I've done one of two things on systems where I've needed to use MeetMe * used a real Digium card, and I've only ever used a TE400 or a TE420 for that purpose, and I know they have the timing chip * used dahdi_dummy, which works well with light load, but I had it running on a very overloaded server and had audio quality issues. I may have had quality issues even with a

Greetings: I'm in the situation where I'm trying to splash information picked off by an asterisk IVR into a Cisco call center environment. I'm under the impression that the ONLY way to do this is to setup socket connections with the Cisco "voice processor", or CVP, and send packets corresponding to GED-125. Cisco has a detailed 100+-page document detailing the internals of

Hello I need to find a recent and neutral comparison of the major products available to connect an Asterisk server to the telephone network, whether ISDN (BRI) or PSTN, and through a PCI card or some external box. I'm told there are less issues (echo, stability) with external boxes compared to PCI cards. Apparently, the main brands are Digium, Sangoma, Rhino Equipment, Patton, and

I run a phone in a remote office using the IAX2 protocol. It mostly works fine; except that every 5 mins it loses connection with Asterisk, before reconnecting 30 seconds later; rinse & repeat. Using the IAX2 debugging, I'm seeing this a lot: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 00018ms SCall: 04050 DCall: 00000

...take me dangerously close to the > upper limit of good call quality. > > Am I complete off? > > Mike > > > -----Original Message----- > > From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- > > bounces at lists.digium.com] On Behalf Of David Backeberg > > Sent: Thursday, March 26, 2009 18:40 > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [asterisk-users] Asterisk multi-cpu > > > > On Thu, Mar 26, 2009 at 3:06 PM, Mike <list at virtutel.ca> wrote: > > > Hi, > > &...