similar to: Server Setup Advice

Hello! I have a sip device that is sending in the SDP: rtpmap:98 g729a It does not seem like Asterisk is negotiating the codec properly, because while the call rings, the rtp lines fail. However, on other sip devices that have "rtpmap:18 g729" in their SDP, things work fine with Digium's commercial g729 license. How do I get "98 g729a" recognized by Asterisk? Thanks,

Hello! What kind of sms text messaging capabilities does Asterisk have? I do not know very much about about SMS technology, but I am looking for the following features: 1. mobile SIP devices can send and receive SMS messages 2. Asterisk server be able to accept and send SMS messages through PRI lines and Internet connections. I noticed that Asterisk has an SMS function, but I am not farmiliar

Hello! I am having an odd problem in that when the caller dials extension "2" in a dialplan, the system waits 3 to 4 seconds before proceeding. This doesn't happen when any other other extensions are dialed, including an identical dialplan on other another extension! Is this a bug? Later, Elliot

Hello! I am trying to configure my digium TE220 dual-span pci express card with Dahdi. I seemed to have managed to set up the card with the Dahdi kernel, as demonstrated by executing dahdi_scan: [1] active=yes alarms=RED description=T2XXP (PCI) Card 0 Span 1 name=TE2/0/1 manufacturer=Digium devicetype=Wildcard TE220 (4th Gen) location=Board ID Switch 0 basechan=1 totchans=31 irq=16

Hello, I have a Google Voice phone number and want to connect it to my asterisk box to have calls handled to my SIP account. When I call the number I receive the correct INCOMING request on Jabber portion of asterisk, but the call is not connected to the gtalk part. JABBER: asterisk INCOMING: <iq from="+ 17174695631 at voice.google.com/srvres-MTAuMTc2LjEwOC4xNjo5ODQ4" to="

Hello! The T.38 Digium Fax Driver sometimes responds with a successful sending of a fax, when in fact, the fax did not go through. 1. Where does this problem lie? 2. How to go about fixing it. Thanks, Elliot

Hello Everyone! According to voip-info.org the correcy syntax for the realtime function is: REALTIME(family|fieldmatch[|value[|delim1[|delim2]]]) on read REALTIME(family|fieldmatch|value|field) on write It seems from the syntax that it is only possible to retrieve a full row according to the value of only of column. This translates in SQL language as Select * from family where fieldmath =

Hello! I have a few sip devices and it is necessary for me to disable call-waiting and immediately return a busy signal if the sip's channel is busy on them. What is the procedure to do so in Asterisk 1.4? Thank you, Elliot -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello! I have a 64 bit Asterisk system and am wondering how to use Digium's 32 bit fax driver. Is there some kind of emulation that can be used? Thanks! Elliot -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090604/b7f1d6c4/attachment.htm

Hello, I am wondering how to send a call to a specific IP address that is different than the host of the URI. For example, an invite to the URI is " john at phone.com" needs to be sent to the IP address 123.456.789.255, not to the IP address of phone.com. How is this done? Thanks, Elliot -------------- next part -------------- An HTML attachment was scrubbed... URL:

Does any have experience with E1 telephony support plus asterisk in costa rica ? Regards, Luis Morales -- --------------------------------------------------------------------------------- Luis Morales Consultor de Tecnologia Cel: +(58)416-4242091 --------------------------------------------------------------------------------- "Empieza por hacer lo necesario, luego lo que es posible... y

Hello! I am wondering how the differences between G729, G729a, and G729b effect call bridging and server interoperability. For example, can one server use the G729 code with another server that uses the G729A codec? Also, which version is Asterisk set up to use? Thanks! Elliot

Hello! I remember a while back I saw a way to answer a call from a device that is not from the one ringing, but I don't remember what how to do it. Any help would be great! Thanks, Elliot

Hello! Does anyone have any experience with installing CentOS 6 (specfically, 6.8), on a Dell T7910? I've tried at least a dozen installs, everything gets configured, and when I have the system reboot, I get 'No boot device found press any key to reboot the machine'. In BIOS, I've enabled AHCI, Legacy boot and modes, and enabled the SAS controller. The disks are seen and

Dear asterisk users, We want setup TE121 digium board: Model: Digium TE121: VoiceBus technology allows the TE121 to use an industry standard bus-mastering PCI Express interface. http://www.digium.com/en/products/digital/te121.php My platform Server: HP Proliant 150 G5 OS: UBUNTU x86_64 GNU/Linux Asterisk: 1.4.21.2 zaptel: SVN-branch-1.4-r4662M When we enable zaptel driver for TE121, the

Hello, Is SS7 and PRI in any way compatible in that if the interface is configured one it will work for the other (granted, it will not have any of the ISUP, etc. parameters available if the line is PRI) or are they two distince protocols that have incompatible signalling? Thanks, Elliot

Hello! I am wondering how to configure Asterisk and devices so I can use different codecs for upstream and downstream packets. Thank you, Elliot

Hello! I have a very simple bash script: #!/bin/bash asterisk -rx "sip show peers" > /var/log/devices When I run it in bash shell, everything works fine, but if I background it (by adding & or using bg), nothing appears in the /var/log/devices file. Any reason for this behavior or help would be great! Thanks, Elliot

Hello, There are some parameters in the ISUP data (coming into the network via SIP-T packets) that need to be translated into SIP headers to be used by asterisk for proper call routing. What gateways are available to accomplish this? Thanks, Elliot -------------- next part -------------- An HTML attachment was scrubbed... URL:

Does any know what happens with svn repository on svn.digium.com ? -- --------------------------------------------------------------------------------- Luis Morales Consultor de Tecnologia Cel: +(58)416-4242091 --------------------------------------------------------------------------------- "Empieza por hacer lo necesario, luego lo que es posible... y de pronto estar?s haciendo lo