search for: gradwel

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 to rgadwell I'...

...om 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 to be an authentication isse. The only support Gradwell offer is for users of AAH, this is a raw Asterisk install I'm working with and their AAH examples don't work. Gradwell did work fine in the AAH test system. If anyone has any working configs could you let me see them? (on or offlist is fine). Thanks in advance. -- Robert P. McKenzie...

...now also use 112 which is consistent with continental Europe). I can't find a call placed at the relevant time that had these numbers, even as mid-part of a string. Below is the part which deals with our external calls. As you can see, calls are routed out via zap, or VOIP (that's the gradwell bit). If someone prefixes a call with "9" it forces it our via VOIP and if someone dials "999" it is intercepted and sent via the zap channels. If no zap channel is free, a call on channel 1 is ended and the number re-dialled. This makes sure that emergency calls can always b...

...e 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 stream and maintained the same codec, so that all asterisk did was signalling conversion. sip.conf... --- [sip-router-1.gradwell.net] context=sip-inbound type=peer host=sip-router-1.gradwell.net [sip-router-2.gradwell.net] context=sip-inbound type=peer host=sip-router-2.gradwell.net --- iax.conf... [general] bandwidth=high disallow=lpc10 jitterbuffer=yes dropcount=2 maxjitterbuffer=500 maxexcessbuffer=80 minexcessbuffer...

...;<<<<=".$ani); $agi->set_variable("CALLERID(num)","01133200274"); $ani = $agi->request['agi_callerid']; $agi->noop("My CalleID: <<<<<<<=".$ani); $agi-> exec('Dial',"SIP/00918885268942 at sip.trunk.gradwell.com,60,r"); //$agi-> exec('Dial',"SIP/00918885268942 at voipon,60,r"); ?> And CLI> == Using SIP RTP CoS mark 5 -- Executing [101 at outbound:1] Answer("SIP/2209-000026d3", "") in new stack -- Executing [101 at outbound:2] AGI("SIP...

I've been testing the debug version of AstTAPI, which worked for a few calls, then a bit later in the day (and ever since), when the call is hung up, the TAPI client doesn't get notified. Looking at the server logs, The TAPI message that is sent upon hangup, isn't being sent. exten => h,1,UserEvent(TAPI|TAPIEVENT: LINE_CALLSTATE LINECALLSTATE_IDLE) This is in the same context as

...X connection, but per call - rewriting call numbers and keeping track of connection status. It's going to be optimized for speed - doesn't do any other modification or audiostream translation - only message passing. If someone's interested -- code + short doc is available at http://www.gradwell.com/tmp/iax_proxy.tar.gz Development will continue - any opinions / comments / contributions are appreciated. Stanis?aw Pitucha Gradwell Dot Com

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 readable by the user running rsync. Oh, this is rsync 3.0...

Hi All, I'm trying to debug DTMF issues I have with certain endpoint conferencing systems (external, 3rd party). On our A*k server I log DTMF, and I see that coming through in the log. What I'd like to see is what is sent onto our VoIP carrier over SIP. I can do a tcpdump of the packets, but what am I then looking for? Would it be in the RTP audio stream or within the SIP

...ifically, 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 - the SIP client dials on O2 mobile, call goes out to A*1. - A*1 Dials out to A*k2 as A*k2 is the gateway to PSTN providers and normal office phones. - A*k2 dials some local Cisco phones, then on no answer plays an audio file, so call is ANSWER...

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...e 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 'default'. Any ideas? -- Kind regards, Stanis?aw Pitucha, Gradwell Voip Engineer T: 01225 800 831 | F: 01225 800 801 | E: stan at gradwell.net | www.gradwell.com Gradwell ? Internet for Business People Phone Services | Business Broadband | Email & Website Hosting Can switching to VoIP today put some change in your pocket? Registered Address: 26 Cheltenham...

...er 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 have publically posted, but I can't even get SIPgate to work with this either !! (Can't pass these directly to Gradwell as their SIP trunks don't support it..) A.

...at and qualify for the peer that has problems - rtp comfort noise is simply not sent. After trying to make it work for a day or so, I reported it as a bug (https://issues.asterisk.org/view.php?id=15466) but maybe someone here has some ideas how to make it work? -- Kind regards, Stanis?aw Pitucha, Gradwell Voip Engineer T: 01225 800 851 | F: 01225 800 801 | E: stan at gradwell.net | www.gradwell.com Gradwell - Internet for Business People Phone Services | Business Broadband | Email & Website Hosting Can switching to VoIP today put some change in your pocket? Registered Address: 26 Cheltenham...

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...u use any workarounds? I'm asking here, because it would be strange if that functionality was broken since 1.4.8 and noone noticed ;) Exact scenario I'm using is described in the bug: https://issues.asterisk.org/view.php?id=15833 Thanks for any help. -- Kind regards, Stanis?aw Pitucha, Gradwell Voip Engineer T: 01225 800 831 | F: 01225 800 801 | E: stan at gradwell.net | www.gradwell.com Gradwell ? Internet for Business People Phone Services | Business Broadband | Email & Website Hosting Can switching to VoIP today put some change in your pocket? Registered Address: 26 Cheltenham...

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 will work it out, but right now I just need a reliable provider that I can port a number to. I'm not especially price sensitive, r...

Hi All, I've a single 1.2.17 Asterisk system. Gradwell here in the UK is used for PSTN calls via IAX2. Our 'net link is a dedicated 2Mb fibre connection (of which we have ever used 50% max bandwidth). We've no E1/T1 links, everything is IP based. My boss complains that many of the calls he holds with others has a bad quality. He also says t...

Hi All, If I had a large block of code, eg: [outgoing-pstn-gradwell] ; the caller ID convertion assumes that the last two digits of the callers id ; are mapped to the last two digits of the PSTN number. exten => _0.,1,ExecIF($["${RECORDOUTBOUND}"="TRUE"],Monitor,wav|${TIMESTAMP}-${CA LLERID(num)}-${EXTEN}-${UNIQUEID}.WAV) exten => _0.,2,...

I am considering using iax.cc (sixtel) and wondering if anyone had opinions, good or bad. Are there outages with any regularity? How responsive are tech support? How is packet loss? I am particularly interested in termination to the UK, but will accept any comments people have. Thanks -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US