similar to: long return times from System() calls with 1.6.2.6?

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

Yes, I purchased licenses for Fax for Asterisk and yes I called tech support and had the WORST experience I have ever had with any technical support call. I am running Asterisk 1.6.2.6 and: FAX For Asterisk Components: Applications: 1.6.2.0_1.2.0 voipgw01Digium FAX Driver: 1.6.2.0_1.2.0 (optimized for c3_2_32) The guy was arrogant and absolutely a jerk and I don't like to call

Hi all, I'm running into a easily replicated problem at the moment, with Asterisk 1.6.0.28 (built from source, no special configure parameters, other than a path) running on top of a fully up-to-date CentOS 5.5, and I'm looking for suggestions as to why this is occurring. I've spent some time looking into the issue, and really haven't been able to come up with much. We have the

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:

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

Hello I didn't find the answer in the ATOF 2nd Ed: When using the Record() application, I need to know how it ended: Did the user leave a message, or did he hang up? If the latter, Asterisk stops right there, while I need to run some other commands before hanging up: ======== exten => _[1-4],n,Playback(/root/asterisk_sound_files/leave_msg) exten =>

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

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

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

Formerly on a thread called [asterisk-dev] Where to find the code of application Bridge On Wed, May 6, 2009 at 7:38 AM, Tzafrir Cohen <tzafrir.cohen at xorcom.com> wrote: >> Can someone please tell me in which file the code for the application to >> be found? I was not able to find a file named app_bridge.c in the folder >> apps. > > app_bridge.c ? app_confbridge.c ?

Can anyone recommend some kind of virtual t.38 fax software? I'd like to test/debug some of the t.38 stuff, but it'd be much easier if I had a software client that could just generate the faxes from a workstation, rather than having to sit with the fax machine + t.38 ata to source faxes from. There doesn't seem to be much out there, and the stuff that's out there is kind of

Hi List, i have to put an * between two other SIP gateways and due to some circumstances, i have to use sip over tcp. With 1.6.2.6 this is working fine: sip gw A (deverto4) sends the call, i hand it over to sip gw B (ocs) and that's about it. In the other direction however (ocs -> me -> deverto4) the call setup is complete but there is no audio. I can see the audio in the form of

I can't seem to find a definitive answer on this, and I really don't want to risk breaking a production server to find out; so I am going to try asking this here, and maybe anyone else in the same situation searching the archives sometime in future will find the answer I get. Can you use variables in the target of a GoTo() statement? What I am specifically thinking of is this;

Aloha, We are looking to roll a solution that will have the following network layout: ISDN-PRI <--> Asterisk <--> T.38 <--> ATA <--> Fax Does version 1.8 with the Digium fax driver have this capability? I like 1.8 because it is a long term support version. What ATA's are people using? Any working solutions would be great! Aloha, Matt

Hello, yesterday one of the extensions on my asterisk server got compromised by brute-force attack. The attacker used it to try pull an identity theft scam playing a recording from a bank "your account has been blocked due to unusual activity, please call this number..." Attacker managed to make lots of calls for around 8 hours before I detected it and changed the password for that

Hello my friends... We are having some problems with the fax in our asterisk server... We have: Asterisk 1.4.21.2 Zaptel Version: 1.4.11 libpri version: 1.4.5 Digium Card TDM 410P This digium card has 3 FXO ports and 1 FXS port where we have a fax machine connected! The problem is that we can receive fax very good, but we can't make any outbound fax call, in fact, our asterisk get freezed