similar to: X100P problem with PSTN from BOLIVIA

Hello I've been using VoIP over a not so reliable net: I usually get a 5% to 10% packet loss and a very high jitter. I tried several codecs and parameters, and the only thing left to test is PLC (Packet Loss Cancellement). Have the astesrisk and digium people implemented PLC?, Are they implmementing it now? and, if not, Where can i find an implementation? Thanks in advance -- Jorge

Hi I' ve just installed TE410P and asterisk-0.7.2 from tar.gz on fedora core 1. When i start asterisk it shows me this: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: ast_pickup_call Apr 19 16:52:32 WARNING[-1085304704]: loader.c:358 load_modules: Loading module chan_zap.so failed! Where do i look, how can i debug? Thanks in advance Jorge Verastegui G RedCetus S.R.L

Hi i have buy two license of G729 codec, but after run Registration program i notice this error Registration unsuccessful... Error code: 110 ERROR! Your Internet connection is probably behind a proxy and the Registration program can't communicate with our server, however it has created the file: /var/lib/va-infoclient which contains your machine signature and

Hi, I have a serious problem to configure Cisco AS5XXX and Asterisk , I trying to use asterisk for PSTN(A)----> Cisco AS5xxx ----> ASterisk---->PSTN(B) (No Nat, no Firewall) I hear (on the PSTN(A)) clearly what the other person is saying, but the other person (on the PSTN(B) side) hears nothing from PSTN(A). I use tcpdump for debug de rtp trafic, and ouput contains

Hi folks, where can I find the R2 beta code for Asterisk? Best, PauloHM -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030806/9c7a0660/attachment.htm

Hi, I've connected a TE110P from digium with a 2 E1 to a Siemens PBX using asterisk from ubuntu linux. Everything is working as expected. This box is being used as a H323 gateway to the pstn. There are few complains but it is working pretty well overall. There is one thing that is bothering me. Asterisk says: May 22 05:03:39 WARNING[9360]: PRI: !! No channel map, no channel, and no ds1?

Hi I'm buying a new box and it brings the new PCI-e standard and not the old PCI slots. I would like to know if the Digium Wildcard TDM400P and <http://www.digium.com/index.php?menu=wildcard_tdm400p2>Wildcard TE405P will work with this PCI Express. <http://www.digium.com/index.php?menu=wildcard_te405p> Has anybody worked with PCI-e yet? As far as i understand, the Wildcard

Folks, I'm trying to get Asterisk to load my voicemail configuration from MySQL. I've followed the instructions at: http://www.voip-info.org/wiki-Asterisk+voicemail+database I restarted Asterisk, but no luck: the voicemail.conf does not get updated. I started with a sample voicemail.conf that I found on the Wiki. Or was it from Voicepulse? I can't remember. For initial testing, I

Hi, I've connected a two T100P from digium with a 2 Rhino channelBank. Everything is working as expected. but I have occasional Falls, asterisk take 99% of CPU resources, with the following report May 24 14:52:41 WARNING[11441]: We're Zap/1-1, not H8?F??n** May 24 14:52:41 WARNING[11441]: We're Zap/1-1, not H8?F??n** May 24 14:52:41 WARNING[11441]: We're Zap/1-1, not H8?F??n**

If you set "dialout=fromvm" in your voicemail.conf, how do you then go about being able to dial back out? Is there a service feature code?

chan_sip.c6524 reload_config= unable to get ip address from asterisk, sip disabled The ip address is working fine, Internet works great. Can anyone help...Thanks

I have the following problem: I have installed two T1 digium card (old T100P cards), plus a TDM400 with 4 fxo modules. Several times in the week I have thousands of warnings like these in the log Aug 1 08:54:47 WARNING[2243]: We're Zap/18-1, not G\uffff\uffffG Aug 1 08:54:47 WARNING[2243]: We're Zap/18-1, not G\uffff\uffffG Aug 1 08:54:47 WARNING[2243]: We're Zap/18-1, not

I have "externnotify = /var/lib/asterisk/agi-bin/vm_notify.pl" so that when someone is left a voicemail it will call the person's mobile phone and prompt them with the new message. The perl script simply originates a call to a persons mobile phone and connects it to their voicemail using VoiceMailMain. Problem is when user hangs up from checking their messages, it runs the

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I have Asterisk sitting between the PSTN and a legacy PBX. Asterisk is doing some IVR work prior to forwarding calls to the PBX and it also acts as the voice mail server for the PBX, with Asterisk configured for IMAP storage. When a call comes in and the caller leaves a voice mail, the VoiceMail application calls the program configured in

Not sure if this is a bug yet, so I wanted to ask around to see if anyone else was having this issue. I have pollmailboxes=yes set in voicemail.conf but externnotify is not called. I know it isn't the externnotify script because if the changes are done in asterisk then it is called properly, if the changes are done via our webserver then it is not. Also, we use odbc voicemail storage. Thanks

I have IMAP voicemail working with Exchange 2003 using a single username and password for multiple mailboxes. Right now, I am setting up asterisk to use voicemail with my Cisco Call Manager (Which I detest BTW...) and I have everything working, EXCEPT: I cannot get my externnotify script to run when any changes have been made to the VoiceMail... Scenario: Bob gets a call -> Bob

I'm trying to use Asterisk for its voicemail capabilities while interfacing with a legacy Toshiba PBX. Is there a way to have Asterisk send a DTMF code to an extension to turn on the message waiting indicator light? When a user leaves a voicemail, I want Asterisk to pick up one of the lines attached to it, and then dial #63<ext>, which is what sets the message waiting indicator light

Would anybody know how to run a perl script as a daemon that would stay connected to asterisk via AMI? Right now, my AMI script connects to the manager interface, originates a call, disconnects. The script will be run maybe 20+ per minute. It would make more sense to me to have the script run as a daemon and have a persistent connection to asterisk's AMI. Thank you in advance for your input.

What's wrong with ARA (asterisk realtime architecture) from voip-info: Asterisk, SER and MWI http://mail.iptel.org/pipermail/serusers/2004-December/013727.html Actually I wrote a patch for this and it supports ast_data too. What you do is tell asterisk that all of your phones IP addresses are your SER machine. Then when a message gets left Asterisk sends the NOTIFY to username at

Is there a way for asterisk's voicemail to send an email (including voicemail attachment) to multiple email addresses? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110408/8908db5f/attachment.htm>