search for: elvish

...0 asterisk systems (about three extensions per site) so the operator can immediately see any extensions that successfully initiate a call. Any information would be greatly appreciated. Kind Regards Stuart -------------- next part -------------- A non-text attachment was scrubbed... Name: stuart.elvish.vcf Type: text/x-vcard Size: 385 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060326/bda46fd6/stuart.elvish.vcf

...le of different codecs (g729 and ulaw) and the problem seems to still exist. The problem happens when the ATA is both internal and external to the VoIP server network. Does anyone have any suggestions? -------------- next part -------------- A non-text attachment was scrubbed... Name: stuart.elvish.vcf Type: text/x-vcard Size: 385 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060525/6f370b42/stuart.elvish.vcf

Hi, I was wondering if anyone else has a similar setup and can suggest settings for the jitterbuffer: I have a client with an ADSL connection at site A & B with site A being dedicated to voice and having no Asterisk server, site B combining voice and data with traffic shaping and housing an Asterisk server. There seems to be packet loss / jitter on this connection and I wanted to know

Hi, I have just completed the deployment of a couple of Grandstream phones (for internal IP use) and was wondering how much harder it would be to deploy a Polycom IP 300 phone. The Grandstream was quite easy to deploy and gives us good voice quality over DSL, however from some of the previous posts I am see that some people had troubles with the Polycom 300. The variant I am looking at

...ase is used for CDR's, sip.conf and voicemail.conf but extensions.conf is static. So with all the above information, I am leaning towards the error being related to the database connection for real time and it occurring when an extension re-registers. Any thoughts? Thanks in advance. Stuart Elvish

Hi, all I am in Australia and I have to setup Asterisk in few offices. There will be IP phones in each office and I must be able to call between offices. I need actual handsets. I need "standard" handsets to be used by people. Those must support features like CID, call forward, etc. --- your normal office feature set. Also I need some sort of more complex handset to be used by

Hi, I was just wondering whether or not anybody has a dial plan or some advice on getting a SIP phone to ring without answering the PSTN line so that the caller doesn't have to pay for the phone call unless it actually get answered by a human or the answering machine after 40 seconds. I had a look through the wiki but there wasn't anything I could find (probably the wrong search

...an't place my finger on what is happening but it appears that when the peer de-registers the defaultuser field is cleared. I will be having a more detailed look through the logs and possibly add more logging to the database but wondered if anyone else has had this problem. Kind Regards Stuart Elvish

...but in the lack of ogg@xiph.org or ogg-dev@xiph.org this will have to do. I've been thinking for a few weeks that Ogg needs a simple text stream (read subtitle) format to go along with theora. This is important, because otherwise I can't transcode fellowship of the rings while keeping the elvish-speek, unless I render the text onto the video frame, and that's not cool. As you can see, the world will end if there is not a subtitle format for Ogg soon. This is what I've come up with. Goals: To create a generic text stream format which is flexible enough to be used for subtitles...

Hi all, Does anyone know if it is possible to configure extconfig.conf with multiple DSN's for failover? The current configuration option appears to only support definition of a single DSN with an optional database table. Sample configuration is: sippeers => odbc,mysql-asterisk,sipendpoints I am currently using Asterisk -> ODBC -> MySQL Cluster and would like Asterisk to

Hi all, Just hit this problem for the first time: WARNING[17712] chan_iax2.c: Too much delay in IAX2 calltoken timestamp from address 10.25x.xxx.160 When I ran "iax2 show peers" everything comes up as unreachable, no calls are passed between the servers (as would be expected) but there is no problem with DAHDI / SIP channels. At the same time there are no network issues (can ping all

Hi all, I am currently testing a new version of firmware released by an ATA vendor and I have come across a strange problem in 1.8.9.0. Sometimes if I dial immediately after hanging up a previous call (we used voicemail for our testing as it has "unlimited" capacity) Asterisk will return a 404 code instead of doing the usual INVITE - 401 - INVITE sequence. The CLI says that the call