search for: alayon

Hello all, Has anyone tried 8xE1 in one box using Asterisk and Digium boards ? What is the CPU needed for sustained performance in this capacity ? Is this affected if G.729 codec is used ? Regards, Jorge A.

Hello, I am new to this list and to asterisk and going through the archive file I did not find an answer to my problem. I have a VoIP network working fine with multiple gateways registered to a Cisco H.323 Gatekeeper. I have successfully registered Asterisk as a GW in that network and also successfully registered two X-Lite SIP Client to asterisk that call to each other. I want to connect to

Hello, I am trying to let someone know that is being called from a specified location. For that, the command: exten => _107.,1,Dial(SIP/SIPGW/${EXTEN:3},30,A(Anounce.gsm)) should let the called person hear Anounce.gsm as soon as he/she answers. (Only calls with prefix 107 are given this notice). The call proceeds fine, but no one hears AnounceSPF.gsm. I tried putting this file in every

Does anyone know how to maximize music on hold quality on calls inbound from PSTN? I know that it is common to have choppy and static sounding music on hold when connecting via PSTN but how can that be minimized? I assume that the bitrates, type of music, etc can minimize the effects. Does anyone have any experience in this area? Do you know where I should look for more information?

Hello all, After compiling successfully Asterisk and AMPortal, I cannot make the fax module work. Asterisk does not start (unless I remove the modules or mark them as Noload in modules.conf) with the following error: Jun 3 20:55:25 VERBOSE[3328]: [app_rxfax.so]Jun 3 20:55:25 WARNING[3328]: /usr/local/lib/libspandsp.so.0: undefined symbol: dds_modf Jun 3 20:55:25 WARNING[3328]: Loading

Hello all, Has anybody cofigured in SIP the Audiocodes MP108 FXS in a way that each port is an extension of the Asterisk Box ? So each port can have it's own mailbox, etc ? Regards, Jorge A.

Hello, I am implementing a small call center with 1 to 4 agents. For some reason I don't understand, if an agent is busy, and it is his/her turn in the queue round, asterisk gives an "all destinations are busy" message and hangs up the call. Agents are SIP lines registered with an audiocodes MP108FXS which registers each line independently. Ringing strategy is RoundRobin (most of

...w stack == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'Local/8521@from-internal-8268,2' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'Local/8521@from-internal-8268,2' Any help will be appreciated. Regards, Jorge Alayon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050805/384935d7/attachment.htm

Hello, I am trying a setup that is the following: SIP Phone (Zultys) --> Asterisk ---> H.323 GK (Cisco) ----> PSTN Any calls from H.323 GW through GK goes to PSTN, no problem. SIP Phone registers to Asterisk, and calling to Voice Mail, No Problem. SIP Phone to PSTN, rings normally, on the PSTN, then connects when the PSTN phone picks up, no audio on both directions. PSTN GW support

Hello, I'm trying to use Asterisk as a SIP PBX with H.323 trunk connectivity. Everything works except that calls that comes from the H.323 side do not get audio both ways. Since the other way round works fine (calls to H.323 side), I suspect the problem to be in the way VAD or Silence suppresion is negotiated. Is there a way to disable VAD in the Asterisk for H.323 gatekeeper connectivity ? I

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

Hello, I got 3 Polycom 300 phones, and upgraded to the latest firmware provided by the reseller. This is my first experience with Polycom and I cannot make them register in my Asterisk Box. I followed every advice I found (including separating [user] and [peer] in sip.conf. Using ethereal, I found that it tries to SUBSCRIBE to the asterisk box and it receives a 403 FORBIDDEN message. I