similar to: Asterisk 1.4 reliability problems

What does it take to get ztdummy to work correctly? I have a new laptop HP HDX9200. I am running asterisk 1.4.19 and zaptel 1.4.9.2 Zaptel compiles fine. asterisk compiles fine. ztdummy loads asterisk runs. Problem is playback() does not work. So then I stop zaptel, asterisk runs and playback() now works. However, meetme()'s dont work. I need ztdummy I'm pretty sure for that. I am

Hello all, I've been lobbying for some time at the #asterisk IRC channel. Until now, I still can't find a solution to my one way audio problem. I rebuilt the Asterisk-1.4.21.2 from the Debian Testing repository on my Debian Etch. I got a Digium TDM400P with 1 FXO (channel 4) and 1 FXS (channel 1). My SIP extension phone located inside the LAN is a SNOM 300 IP phone. This one way audio

I'm trying to understand something that just doesn't seem to compute. How can companies like Cisco justify selling their hard phones for as much as they do? I know there is a matter of recouping R&D costs but when you look at the iPhone with all its amazing features for less than $500.00 it just doesn't make sense. Am I the only one that thinks this? Roy Anciso Director of

On Mar 19, 2008, at 1:00 PM, asterisk-users-request at lists.digium.com wrote: > Am I expecting too much? Perhaps. I think the hardware on which we run Asterisk can be much more reliable than the software, which is often the case. We have a bunch of HP servers with RAID and have never lost anything. A HD may fail, but the RAID keeps it going until we pop a new drive in there. A

Hello, can i use different audio codecs when i calling between sip devices ( snom phones ) and different when i making call from isdn to sip or from sip to isdn ? best regards Marian -- SUNTEQ s. r. o. Bojnicka cesta 35 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ ------------------------------------------------------------ A

Hi list! While looking for the Snom 190 I found another phone, the Elmeg IP 290 (www.elmeg.de). Looking at the pictures & the specs they seem to be very similar beasts but the firmware is supposedly not interchangeable. Does anyone know the difference between the 2, do they work with Asterisk? The weird thing is that Elmeg has similar phones with the Snom look but they are ISDN only (no

Hi, i am from Germany, so excuse my School English. I use Asteriks 1.2.12.1, zaptel 1.2.9 and mISDN Rc25 - since my update of Asterisk 2 wooks ago, Echos accure in my SIP Calls. I use SNOM 360, sometimes there is no echo (for example if i call myself via SIP->Asterisk->SIPProvider->TELEKOM->ISDN) but if i call other people there occures Echo many times. The Routing is always the

I've been evaluating Asterisk for a while, and things seem to be going very well. The issue of redundancy and automatic fail-over is now on my mind. I searched the archives and googled for solutions, but didn't really come up with much. We'll be using queues (modified), which precludes some of the standard redundancy solutions, since the queue needs to know all the agents

These taken tonight (9/22/2004) at the Expo and Reception Enjoy. http://photos.tropiano.org/gallery/astricon-2004 Lenny

Hi Everybody I've got a weird problem. I am running one Asterisk system on a dual processor box. This box mostly do VoIP only but it has a Fritz PCI ISDN card installed with latest drivers. Dialing out through the ISDN cards from an internal Snom phone works fine and so does dialing in. Except - if I load the ztdummy module (for IAX channels) the capi drivers starts acting up. It is hard

I want use iLBC and have following in mind, please help me is it possible ? ISDN <-----(ALAW)-----> * <-----(ALAW)-----> SNOM SIP??<-----(iLBC)----->?*?<-----(ALAW)----->?SNOM 1. ISDN incoming codec is ALAW SNOM codec should be ALAW (can't be iLBC because a lack of codec). 2. SIP incoming codec should be iLBC (snom is ALAW). 3. SIP outgoing codec should be iLBC /snom

Hi All, I have setup 2 asterisk servers using AAH 2.8. I have configured a IAX2 trunk between the 2 servers using the guide on dumbme. Trunk is not using register string and no authentication. In my dial plan I have 7XX numers on server B and 6xx numbers on Server A. Calls from my SNOM phones are ok between the extensions on the 2 servers. In server A I have a eicon 4 port BRI card connected.

Hi, Before diving into this, I would very pleased to know if someone could yes or no, successfully compile mISDN 1.1.8 on Lenny (latest RC1 or beta2 version) ? Regards After a fresh install on Lenny, I can reproduce at will : apt-get install build-essential linux-headers-2.6.26-1-686 cd /usr/src wget http://www.misdn.org/downloads/mISDN.tar.gz tar xvf mISDN.tar.gz cd mISDN-1_1_8 make ....

Hi everyone, Ive just installed a bunch of Snom 360s, and now having a NIGHTMARE of problems! Ive got a receponist phone with a extra sidecar on it. And when she gets 2+ calls and puts them on hold, when she goes to transfer them out the calls on hold get merged together. Somehow the calls on hold get merged and not to the extension needed!! Any help on this would be great guys, that would be

I have a pair of snom 360s at a customer and they were giving me Low Memory errors. The distributor suggested updating the firmware. I did that, to the one just below 4.0 (which wasn't released yet). One of the phones is still giving the Low Memory error every 3-4 days. The other one had a broken display that was just RMA'd, so it' hasn't been up long enough to know if the

Hi, I am using a number of snom190 phones, and an asterisk "gateway" server, and recently started experimenting with call transfers. The snom phones provide support for attended and un-attended call transfer, so I would rather use that than call-parking. I have found that un-attended transfer works fine, and that attended transfer works fine if the originating phone call is NON-SIP

Hello, I have been retained by a Building Management Company to install a combined Voice/Data solution for a Tennated Office Space. This space will rent offices, with telephone and internet service to inviduals or small groups of individuals. As fate would have it, the service will be provided in a building where we have a major Pop, with a DS-3 worth of ISDN PRI circuits, 345 megs of

Hi, I am building for a customer an * solution that will use 2 Digium cards. 1 x TE110P (T1 ISDN PRI) 1 x TDM40B (4 analog ports, 2 for faxes, 2 for extensions) The system will be connected to the PSTN through the T1 ISDN PRI interface. All customer extensions will be SIP phones except 2 fax machines that will use the analog FXS ports on the TDM40B card. I have been investigating on this

I want use iLBC and have following in mind, please help me is it possible ? ISDN <-----(ALAW)-----> * <-----(ALAW)-----> SNOM SIP??<-----(iLBC)----->?*?<-----(ALAW)----->?SNOM 1. ISDN incoming codec is ALAW SNOM codec should be ALAW (can't be iLBC because a lack of codec). 2. SIP incoming codec should be iLBC (snom is ALAW). 3. SIP outgoing codec should be iLBC /snom

Gabe Bourque wrote: > Hello Anton Tinchev, > > I'm writing to you in hopes you can answer a few questions regarding > Asterisk/Digium and it's reliability. I saw your posting in the > Asterisk mailing list (Re: [Asterisk-Users] Is Asterisk ready for "real" > use?) and decided to write directly to you. The reason being that you > are one of only a few