similar to: Callerid problems with 1.0.5

I have a couple of DID's through VP Connect and have been having sound quality issues on incoming calls. During the call, the calling parties voice sometimes sound like it is crackling, in other words it is not very crisp. I would liken it to listening to a radio with a blown speaker. This sound defect comes and goes throughout the call. The other person is always audible but it just isn't

If this has been covered before - I appologize. We use some Sipura SPA-2000's with the g711 codec and all seems fine (except for the occasional failure to register errors in my asterisk logs - but I will save that for another post). g711 call quality is on par with our Cisco 7960's. However, when using the g729 codec, the call quality on the Sipura device goes downhill on the PSTN side

Two cvs-head asterisk boxes with iax2 working fine (without register statements). When two calls are placed simultanously from system A -> B and the packets are sniffed on the wire, I see the two calls using two different udp packets. At the top of iax.conf I have trunk=yes and jitterbuffer=yes (at both ends). I was expecting to see both calls handled within a single udp packet, but

I have a couple of DID's with LiveVoip and am having major DTMF issues on incoming calls. I am connecting to them through IAX using ULAW. When someone dials one of these DD's (from a landline) they are for the most part unable to navigate the IVR menu successfuly. I would say the failure rate is greater than 80%. For example if the caller presses 5 sometimes * will see the DTMF as 55 or

I run an automated information retrieval system, using Asterisk. Fairly often the system misses a dialed digit. Our codes are all 4 digits, see lots of logs with: 4199 - OK 530 - Invalid code 330 - Invalid code 5330 - OK As callers experience skipped codes. We're using Broadvoice SIP with inband DTMF (and we've tried every possible setting or option

Hello, Just confirmed this on my end, because of the massive changes that have been made to callerID handling in asterisk 1.0.5 many of the features of the astGUIclient suite will not work on this new version. The latest stable version recommended is Asterisk 1.0.3. We will work on trying to find ways around the new callerID rules that the asterisk developers have put in place and hope to have

When I make a call over the Internet and call myself IN over POTS my phone rings to outside party but I can not hear it. Why isn't my channel extension indicating busy status when I'm making call over Internet? This way I could ring my next extension with n+101 priority. I'm using Sipura-3K unit. -- #Joseph

>> Has anyone benchmarked Asterisk on a dedicated single versus dual >> processor machine? > > http://www.astertest.com/ > > Cheers, Philipp The test results that Philipp pointed out show some protocol comparisons that include "iax2 trunking / alaw" and "iax2 / alaw" and concludes that "IAX2 trunking is more than twice as fast as non trunking

Hi all, I use a GSM device to send dtmf on my asterisk system (via SIP). the codec I use is ulaw (or a-law). dtmf mode is INBAND. relaxmode is on. but most of the case, I 'missed' some DTMF or I 'double' one. as anybody as seen this before? is there any way to prevent this thanks

Until recently, I was using Broadvoice for my in/out calling thru Asterisk. I was extremely pleased to see that Broadvoice was actually passing the callerid info (number and text) that I had set up on each device in my SIP.CONF file. I had PSTN users tell me that they were actually seeing name and extension info when I called them from the Asterisk box. Last week, due to numerous user quality

I'm having a very annoying problem with access my asterisk system from work. Our phone system here only produces very very short DTMF tones. The phones work fine for other IVR systems (Dell Support, HP Support, etc, etc). However, tones to Asterisk just never make it. The way I'm calling into my Asterisk server is such: OFFICE PHONE => CALLUK.COM 0870 => IAX Inbound The

Need to replace our older soho fax machine with something more current. Would like to run the fax line through *, but haven't been able to make spandsp work correctly with digium TDM04b card. Our fax volume is very low (maybe a few per week), but we have multiple offices in three geographic locations and would like to be able to email the images to the correct location. For planning purposes,

Hello! I want to upgrade the docevot-installation from v 1.0.5 to 2.1..7 Now I get the following executing doveconf -n -c /etc/dovecot/dovecot.conf > /home/mail1/dovecot-2.conf doveconf: Warning: Obsolete setting in /etc/dovecot/dovecot.conf:217: add auth_ prefix to all settings inside auth {} and remove the auth {} section completely doveconf: Warning: Obsolete setting in

Hi, I am thinking of signing up with voice pulse connect to connect to my asterisk server and using it as a regular line. Is it good? Or should I go with vonage or others ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050124/16792f10/attachment.htm

Can someone give me a clue as to why I keep hearing DTMF type beeps on my phone calls. It sounds exactly like someone on the other end is pushing a key on their phone but they are not! Has anyone ever heard of this before? It use to happen once in a while, today it's been happening a LOT and it's driving me batty.. -- Start Your Own ISP! http://www.YourOwnISP.com

Does anyone have any experience with iax.cc/sixtel? Are they a legitimate company? From their website it looks like you can get a private incoming 800 number for 30 cents/month plus 2 cents/minute. Somehow that pricing seems a little cheap for a DID number. I assume there has to be some minimum usage or something. Any info as far as actual costs and/or voice quality would be appreciated.

Hi I've taken apart a 7960 to fit a backlight to the LCD. Would others on the list be interested in this as a project when I've finished (i.e. should I document and photo all the stages)? Also I can source some 7960's at a very good price but what are the rules about posting this information on the list if others were interested? P

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I have a stock market trading values time series. What's the best way to extract the "last day of month" values. I looked at function window() but doesn't appear suitable for this since it expects regular dates. Thank you. lukas - -- Lukas Kubin lukas.kubin at permonik.com phone: 00420603836180 -----BEGIN PGP SIGNATURE-----

Hi, I''m trying to stub File.open and whenever I do, rspec blows up with the following error message (undefined method `add'' for nil:NilClass), which seems to happen because method __add in class Proxy is calling #add on global $rspec_mocks, which in turn is nil. Can someone explain what I''m doing wrong, as I can''t seem to stub anything out! Here''s

Hi, I am trying to install node_openstack module for puppet. But I see its dependency on puppetlabs-cloud_provisioner<https://forge.puppetlabs.com/puppetlabs-cloud_provisioner> (>= 1.0.2) Is this version available, if not is there any work around? regards, Haynes Davis ________________________________ This message is for the designated recipient only and may contain privileged,