similar to: Critical Problem with asterisk

Hi all, I m in a trouble using festival voices in asterisk. I am not able to change the default male voice of festival. Although i downloaded the us1 female voice and it iw working good in festival's CLI but it is not coming when i am usinf Festival in asterisk. I changed the default-voice-priority list directive and set us1_mbrole as first entry and also changed voice in festival CLI. But

Hi, I am using asterisk 1.2.1 for building an ippbx for my setup. I am having problem in accessing advanced options option that comes on pressing 5. It says to leave a msg and asks for extension. Now when i dial the extension the IVR silently goes into the top IVR menu. And the asterisk console show the message: "No entry for <EXTEN> in voicemail config file" As i

I am using asterisk-1.2.1-15 and want to use it to replace my normal PBX with it. For creating IVR menus i tried festival, the text which was passed into it was said, but the problem was at stating of every line a "tick" sound comes. As festival app in asterisk connects with festval server at each line(as it required festival_server for its functioning and logs can be seen on

HAllo! This is Lokesh. I want to use speex codec in my project for encryption in GSM phones.Is it possible to use the availabale API in SPEEX in eclipse,i.e java IDE tool.Or else do u hav already available compatible package for java.Eagerly waiting for your reply.Please suggest me how to use speex in my project.Desperately waiting for your reply.Thanks for your time. regards,

Hi All, I am porting the vorbis encoder to Symbian platform,(to support ogg recording) everything works fine for the emulator(PC), but then on the Hardware whenever I am trying to record it gives KErrOverFlow(buffer OverFlow). We get this error from the driver when the latency of the codec is too high. The main reason is that the hardware does not have the floatpoint unit and the

Hi As per the Linphone, Readme.arm I tried to compile the speex. -------------------------------------readme.arm-------------------------------------------------- ........... Cross compiling speex for ARM: ******************************** First you need to remove ogg headers from your build system to avoid a dirty conflict between your build machine binaries and the arm binaries. They

Folks, I have a NEC 2400 pbx(non-voip) behind a Cisco 3725, connected via standard wic-t1 card. The NEC needs to call two different asterisk servers with 4 digits. I have two way calling working with the one * box, but the other is perplexing me. Here's the layout * <--> Cisco 2811(192.168.13.1) <--> 1.54 point to point <- Cisco 3725(192.168.8.1)<-> NEC 2400. The

Hi i test a new equipment on my backbone: a Cisco AS5300 with voice dsp ressource connected at a E1 Voice Link. I want that all call incoming on the cisco 5300 are sent to Asterisk and all Asterisk outgoing call are sent to Cisco AS5300. Actually, i configure the AS5300: isdn switch-type primary-net5 ! voice service voip sip ! voice class codec 400 codec preference 1 g711alaw codec

Setup: Asterisk server in NY. Cisco 7960 IP Phones in NY and London. Dedicated T1 from NY to Ldn. T1: Latency - 100ms Qos applied No errors Default codec on Ldn IP Phones = g711alaw Default codec on NY IP Phones = g711ulaw Both codecs allowed on each phone. Issue: Calls on IP Phones from NY to London hear clicking noise on NY end. Anyone experienced something similar or can offer some

Hi all, Firsts of all, let me say that I'm new to asterisk. I have some time suscribed to the list reading a lot of your messages and trying to learn a lot. The case is: last week I installed an asterisk server in the following scenario: PBX --- CISCO_ROUTER ---- ASTERISK The calls that are routed within the asterisk work perfect, there is not problem. However, the calls that are

Hello; I have a SIP trunk with a service provider, the caller from landline or mobile is hearing us very well, but the agent who is sitting on the handset is not hearing well, the voice at the agent is not crystal (like he is talking from well or far deep place). Although the IP Phones are cisco 7942G and the used codec is g711ulaw (actually it gave better quality than g711alaw). If we increase

We appear to be having a problem with our asterisk setup. We have a cisco AS5300 with pri lines coming in and passing the calls onto asterisk then too the sip phones. the phone call from the sip phones (7960's) appears to be ok nice and clear including the user who has called in. but if your the user who has called in its all crackley sounds really bad when they speak. i believe this

Hello, Has anyone experienced a segmentation fault in asterisk for using G729 against an AS5300 in SIP ? I'm having this problem. Also, any other codec I use gives me incompatible media (can be read in SIP DEBUG messages). AS5300 configured for multiple codecs, so is Asterisk. Tried G711u/A G723 and G.729. Any clues ? Regards, Jorge A. Info: Asterisk ver 1.0.7 stable Using AMPortal

Hi All, I was able to insert some extensions in Mysql DB and use them successfully. In Mysql extensions table the priority column is of type tinyint and when I give 's' value for it, it is not accepting that value as it takes only tinyints. Please tell how can I make that column accept values like t,s,i and make it work with asterisk in realtime without any problem? If I change the type

HI, I have been pulling my hair out trying to get a Cisco MC3810 to interface my Asterisk box with a T1. I am able to make outgoing calls but incoing calls never reach my Asterisk box. The cisco give a fast busy when I try to call one of the DID's. When playing around with the dial-peers I can get the cisco to pick up the call, but then it forwards the call back to the ANI that is dialing.

Hello, I have a cisco ATA 188 registering both of its lines to * I can place calls between then an to kphone an MSN messenger (both registering with * too), a few days ago a friend lend me a Cisco IAD 2430 and I was willing to do the same thing with it, since it has 24 ports I was willing to to use 24 analog phones with it however something really weird happens I can place calls from my ata,

Hello, I've got an ISDN PRI circuit terminating in a Cisco AS5350, which in turn is talking to an Asterisk server via SIP for call origination and termination. Seems simple enough, and it works for the most part, but: 1) Caller ID name data comes in on the PRI, but doesn't appear to get handed off to the Asterisk server via SIP, at least not in any format that Asterisk

All, I need to run this by everyone and see if someone has any idea's. I have a asterisk server setup and currently am receiving the inbound calling number where the name should be. My setup is.... One pri terminating into a Cisco 2431 router Sip messages from the Cisco get sent to a asterisk server linksys ata's a each remote end. I can receive the calling name if the call originates

Hello, everyone. Right now, we are trying launch our own PBX system based on Asterisk(Fedora) with Cisco 2611. Cisco has 2 port FXO card installed on it. For testing, I have 2611 hooked into phone line with number of xxx-xxx-xxxx fine. (I'll call it F). Using softphone, I can dial in extension 1001 on asterisk, which should talk to cisco. After initial connection to Asterisk, I have try to

ala cisco 7960 -----Original Message----- From: Matt Schulte Sent: Thursday, December 16, 2004 10:34 AM To: 'Paul A Brown' Subject: RE: [Asterisk-Users] Cisco 7960 (SIP) hold problems Sure thing, the biggest problem I had was getting the SIP filenames working correctly for updating the firmware (blah, I love Cisco but these phones are a joke for support). This works for me! Good luck.