similar to: Codename Pineapple - Chan_sip3 - what's the status?

Friends in the Asterisk community, I've been talking for years about the new version of the SIP channel. I've been trying to get funding and get going. Well, the funding part remains to be handled, but I have other news - if you kan keep it to yourself. ...I've began coding. Finally. With a happy smile on my face I removed "pedantic=yes" the other day. After years of

The Asterisk Developer Team is proud to announce the Asterisk SPE v1.0 Beta Release for immediate download on tftp.digium.com. The SPE has been developed as a joint project between Digium, the Asterisk Company, Voop, the European Asterisk Dialtone provider and the Asterisk community. The Asterisk Service Provider Edition is focused on the needs for the new breed of Telecom companies - the

Friends, Beginning next week, I will travel around Europe to teach Asterisk - the one day Meet Asterisk training. MeetAsterisk is organized by Edvina in cooperation with Digium and Voop. In many places, local Asterisk equipment resellers participate and show their equipment. This is the tour plan: * Amsterdam April 26 * Copenhagen April 27 * Oslo April 28 * Paris May 3 * Brussels May 4 *

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 <p><p>re all! about the running question (as i read on the http://icecast.org website in the 3rd party software section) if MuSE is working with icecast2, here is the answer: it definitely does :) the problem was fixed a couple of months ago in the CVS code, now we have a new stable release which will get hopefully soon properly packaged

Hi folks - I've recently returned to asterisk after an eighteen month break. I've two sip providers - gradwell in the UK (inbound and outbound) and talklite in the US (outbound only). I've managed to get outbound dialing working but am not receiving any calls from gradwell. I've included my sip.conf and extensions.conf as well as the output from tethereal. When a call is placed

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 annunciazio' annunciazio'! dyne.org autoproduzioni & the FreakNet Medialab proudly present: __ __ ____ _____ ___ ___ _ | \/ |_ _/ ___|| ____| / _ \ / _ \ / | | |\/| | | | \___ \| _| | | | | (_) || | | | | | |_| |___) | |___ | |_|

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hi We take calls inbound via SIP from our Cisco PSTN gateways, and pass it to customers using IAX (they run their own asterisk servers). We've noticed that asterisk is transcoding the call into a different codec, if the customer prefers a codec different to that which our cisco gw prefers. As such, the quality of the call can degrade. We'd rather asterisk just passed through the RTP

Hi Amit OK, the majority of our calls go out via zaptel fxo and pstn lines. When these are all busy, calls are routed via a VOIP provider here in the UK. All activity is recorded in our logs, and I can find no trace of either 999 or 112 (if since been reminded that in the UK, you can now also use 112 which is consistent with continental Europe). I can't find a call placed at the relevant

Does anyone on-list have an example of working SIP config using Nuvio? I have a byod account that I'd like to run into my Asterisk server. Thanks, Michael

Hi, I've got different customers that may use the same asterisk. Each user can blind-transfer a call to whatever place they want. But of course the transferring side should be billed for it. What can I do to see the difference between the channels here? If there is an A->B call going on, I'd like to know which side did the transfer - but whichever side does it, I get back to context

I'm getting this error message and I don't really understand what rsync is trying to tell me:- rsync: link_stat "/rdiffBackup/gradwell/Mail/." failed: No such file or directory (2) rsync error: some files could not be transferred (code 23) at main.c(977) [sender=2.6.9] Can anyone explain what it's saying please. /rdiffBackup/gradwell/Mail/ does exist and is

Hi, I've got a problem with rtp keepalives. I'm using basically the same config on 2 hosts, but one of them sends rtp comfort noise when it's on hold, the other doesn't. The only difference I can think of now is that one of the machines is multihomed, but that might be unrelated. rtpkeepalive is set to 2 and I can confirm is by doing `sip show settings`. I've tried all

Hi, I'm having problems with sip remote attended transfer using 2 asterisk boxes (same version, latest 1.4.X). Whenever I transfer from a call from box A to a call on box B, one call leg of the transferring phone is not disconnected (the one that is normally dropped by server side, phone disconnects the other one). The same situation works perfectly with local attended transfer. Is anyone

Does anyone here use either Gradewell or inWeb for service? They are both UK based. I'm trying to get a couple of inbound IAX2 based numbers from both of them to work with no luck at all. The one thing that sets these guys apart from the rest of companies offering inbound numbers is they tie the account to the IP of the asterisk server. Neither use register lines in iax.conf, there appears

Hi All, I'm looking for how to enable SIP Markers, or specifically, how to have the TIME reset when a call route changes. I'm debugging an issue, where a sip client we have switching to one-way-audio, when an asterisk server fruther down the call path dials out to the PSTN. Scenario is: SIP Client -> A*k1 -> A*k2 -> PSTN Provider/Gradwell -> O2 -> Mobile

Hi, I am using phpagi for agi scripting. and want to update callerid number but didn't get any success. please help me how to update PHPAGI is new for me. Below is the code which I write. #!/usr/bin/php -q <?php set_time_limit(30); //require(.phpagi.php.); include("phpagi.php"); $agi = new AGI(); //answer the call $agi-> answer();

Hi list I've written a tool that works as a lightweight (standalone - no asterisk) balancer for IAX servers. It's in early development now, but seems to be stable enough and handles couple hundred simultaneous calls with not much latency (SIPp + asterisks tested). It's configurable by listing servers' IPs in iaxproxy-servers file loaded at startup and will keep track of load on

All, I've been experimenting with shortcodes for SIPgate etc. Passing calls to SIPbroker seems a good way to go, but the message I've had back from SIPgate is "we don't support SIPBroker"... So whats the easiest way to support SIP <> SIP network calling? At the moment, I've setup some local shortcodes (eg dial **777. to goto sipgate.co.uk) based on what Gradwell

Hi, I realise this is probably the wrong list for such a question, but I need a pointer to somewhere I can get some feedback on experience of (business class) voip providers for the UK? Situation is that we are currently with Gradwell and use them for an inbound/outbound single line for a business and their quality has gone from excellent to abysmal in the last few weeks. I'm sure they