similar to: Change Packetization Time

Hi List, I am trying to call from my asterisk box (1.2.18) to and audiocodes MP114. I keep getting an error from asterisk of -- Got SIP response 415 "Unsupported Media Type" back from XXX.XXX.XX.XX. Both box's are set up to use G729. Anyone have a hint as to what it may be ? Thanks. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL:

I thought I would give the new IMAP support a spin on my home server, but without much luck so far. Asterisk 1.4.0 Dovecot 0.99.14 Maildir format C-client 2006d The imap server is also the Asterisk server, so connections are on the localhost. The error posted to the logs is: IMAP Error: Can't open mailbox {127.0.0.1:143/imap/authuser=root//user=dan_austin}INBOX: invalid remote specification

Hello All, I always used a compiled version for a x86 system >From http://kvin.lv/pub/Linux/Asterisk/ , and Works fine. At this time , I'm using a Pentium IV Dual core, Running Centos 4.3. I tried to install the 64 bits Compiled version but has a translation time > 20ms. Is it correct ? I also tried to make my self compilation and apply The patch but the patch doesn't work on ICC

Hi all, I need to be able to adjust packet sizes and found the patch at http://bugs.digium.com/view.php?id=5162 Thus, I checked out and compiled http://svn.digium.com/view/asterisk-old/team/group/5162_rtp_packetization I added the line "packetization = 30" for one peer in my sip.conf and started asterisk with the "-I" switch for async RTP. That's all it takes

Well, I don't know how to tune it more, it connects at about that rate in a mediocre rural landline. ILBC uses samples of 30ms, so if you set the trunkfreq set to 20 you will be using more of the necesary scarce bandwidth AND dropping sample info in each frame, thus making audio choppy and unclear. Make shure to disallow all codecs and then allow only ILBC or lpc10 (search for it in

Forgive this (possibly) silly question, but my upstream provider requires a packetization of 20ms. Using asterisk-oh323, I can set the "number of frames per RTP packet". How does this equate to packetization in ms?

To my knowledge, Asterisk's packetization rate is hard coded at 30ms. If I wanted to, where in the code could I go to change it to 20ms. Is there anything bad that might happen if I change it (asterisk related)? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070314/6088397a/attachment.htm

> Page 3: > > To be compliant with this specification, implementations MUST support > 8 kHz sampling rate (narrowband)" and SHOULD support 8 kbps bitrate. > The sampling rate MUST be 8, 16 or 32 kHz. > > There is a type above after (narrowband), there is a " extra character. > > I don't understand what is the motivation to specify "SHOULD

Does anybody knows if it's posible or if there is some develoment in course to be able to use longer transmit packet sizes (as long as I know this is fixed in 20ms now) with the compressed voip codecs in asterisk (g729, g726, gsm, etc). I need to use asterisk to connect remote sip clients with 24kb bandwidth lines and I'm using a licences g729 codec but because I can't increase

Hi list, Can anyone who has successfully ran asterisk on a home router please give me the modell number as well as how they did it ? Thanks. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061207/92c7f946/attachment.htm

And it runs linux. http://www.zip4x4.com/ZIP4x4.htm Anyone seen one? bkw

Dear, is any way to change , the size of voice packets? I want to increase the quality by decreasing the size of each packets, because of bandwidth failure. ? thanks in advance Mani -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081110/c1b2ed9d/attachment.htm

I have an installation that connects in a [very] good day at 22kbps, but the normal is about 18kbps. I use de ILBC codec, and also change in iax.conf the trunkfreq = 20 to trunkfreq = 30 It works, you can understand well the other person, but don't expect miracles or an outstanding sound quality. > Dear Dan; > > Thanks alot for your kindly reply. > > Well, what u advise us

Good morning, I have a Polycom 601 with two side cars. I created a list of contacts in XML and it shows up on the side cars exaclty how I set it up in the XXXXXXXXXXXX-directory.xml file (in the order that I wanted it etc.). However when I hit the directories button and then contact directory I see the list in alphabetical order based on the last name. I want it to show up in this list as well in

I am having major problems with the first run of Snom 360s that rolled out last month. I am working with the US vendor and they in turn are working with Snom but I wanted to see of anyone else was using these or having issues with them. Issues: Speakerphone/Hands Free volume spikes up and down during a call. You have to manually set the volume during every call. This makes it totally unusable.

Does anyone know if there is a way to get a DB or any other means to see if I can see if a call is coming from a cell phone or not. If I am able to see if it is cellular or not is there any way to see aprox. what area the phone is in (I know this wont be simple but would it work if I have an agreeement with the cell phone companies) ? This is for the US. Thanks. Dovid -------------- next part

Hi List, I was thinking the other day that even with T.38 there are still some issues with faxing. I was thinking of a protocol that instead of just sending down the fax tones an ATA or "VOIP fax machine" would get the entire fax convert it into some sort of image and pass it down the line to the receiving end. I got the idea from RFC2833. Yes I know that fax machines send bit by bit and

I use the E-series Nokia phones on my Wireless LAN. The e series have sip agent On 8/20/07, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit >

We've been holding back on this release to coincide with the Asterisk 1.4.0 release. This is mostly a compatibility release, but there are a few new features: * No longer requires register_globals in PHP * Separated code from configuration settings in ./lib/defines.php (hopefully this will make future upgrades easier) * Migrated all database interfaces to PEAR::DB which

Hi List, I am using asterisk 1.2.14 with real time and I am trying to send the email to more than one email address. In that field I put in user1@domain.com;user2@domain.com. When the call goes to VM I see in the CLI: uniqueid => 17 customer_id => 0 context => techmast mailbox => 14 password => 1234 fullname => Sales and Service email => user1@domain.com email =>