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Hello to everybody, I have a problem with the disposition filed that asterisk write in mysql table. What I notice is that for every outbound calls (for example to a mobile phone) I see in disposition field the string "ANSWERED" when I reject the call and also when I really answer the call, while in the variable DIALSTAUS I have the correct status of the call (BUSY, CHANUNAVAIL,

Hi to everybody, I have an spa 3102 where i connected an analog phone (in the fxs port) and the pstn line (in the fxo port). This is my problem: the incoming call doesn't arrive to asterisk. In the spa web page i configured this dialplane: (<:line01@192.168.1.220:5060>) where line01 is the context in sip.conf, 192.168.1.220 is the asterisk ip and 5060 is the asterisk sip port.

Hi to everybody, I need some help in configuration of the spa 3102. I created an account for line 1 (user 208, sip port 5061) correctly registered in asterisk, then i create an account in sip.conf like this: [general] register = line01:pwdsipura:line01@192.168.1.222:5060/095377078 [line01] username = line01 fromuser = line01 secret = pwdsipura host = 192.168.1.222 fromdomain = 192.168.1.222

*Update Jul 2007:* For a T.38 gateway you can use Asterisk 1.4's T.38pass-through support in combination with the new OPAL (Open Phone Abstraction Library) - using t38modem (currently CVS) which now supports SIP (and not just H.323) to terminate T.38 calls. You can also use OPAL and chan_woomera to do essentially the same. Where can you find this t38modem stuff ? Google replies things that

described (stop accepting calls and shut down when all calls have completed). If you don't want to stop accepting calls, but still want to stop Asterisk when there are no active calls, you can use "stop when convenient". The same qualifiers ("gracefully" and "when convenient") can be applied to the "restart" command. Cheers, AR On Dec 10, 2007 7:29 AM,

In https://bugs.llvm.org/show_bug.cgi?id=38011 (see also https://reviews.llvm.org/D48721) a problem was revealed related to llvm.loop metadata. The fault was that clang added the !llvm.loop metadata to branches outside of the loop (not only the loop latch). That was not handled properly by some opt passes (simplifying cfg) since it ended up merging branch instructions with different !llvm.loop

<br> Or even:<br> <br> &nbsp;<a href=3D"http://www.blackbox.com/Catalog/Detail.aspx?cid=3D425,1423= ,1424&amp;mid=3D4946" target=3D"_blank">http://www.blackbox.com/Catalog/Det= ail.aspx?cid=3D425,1423,1424&amp;mid=3D4946</a><br> <br> (same thing from the UK site:)<br> <br> <br> <a

or we need to implement a SIP server to integrate with Asterisk in order to= provide full picture of VOIP system? Thanks. > Date: Wed, 5 Sep 2007 13:30:21 +1000 > From: devraj at gmail.com > To: asterisk-users at lists.digium.com > Subject: Re: [asterisk-users] How to make call from asterisk? >=20 > Helps us help you further, what do you intend to do? >=20 > - Dial

... Thanks, anyway for telling as at least, it reflects your needs. > > > You want NT PtMP and i second that, > not being limited on the asterisk > side is a must in the > telephony ecosystem, since the legacy PABX aren't alwsys easy to > reconfigure. > > _______________________________________________ > -- Bandwidth and Colocation Provided by

<br> Or even:<br> <br> &nbsp;<a href=3D"http://www.blackbox.com/Catalog/Detail.aspx?cid=3D425,142= 3,1424&amp;mid=3D4946" target=3D"_blank">http://www.blackbox.com/Catalog/De= tail.aspx?cid=3D425,1423,1424&amp;mid=3D4946</a><br> <br> (same thing from the UK site:)<br> <br> &nbsp;<a

the Telco, I can make calls in. What I am trying to get though is how to pass through the DID range. The E1 that I am connecting to the Telco with, used to connect direct to the NEC system and already has hunt group calling enabled for all 30 channels. Also, I was given a 100 number indial range (from 00 -> 99). If the E1 is connected to the NEC directly, I can call 5555 7320 and the NEC

asterisk*CLI> dahdi show status Description Alarms IRQ bpviol CRC4 T2XXP (PCI) Card 0 Span 1 OK 0 0 0 T2XXP (PCI) Card 0 Span 2 RED 0 0 0 On Thu, Apr 2, 2009 at 9:40 PM, Martin <asterisklist at callthem.info> wrote: > That's very strange ... the code when is

connecting legacy PBX to Asterisk (for the very same reason, those PBX use TE-PTMP). If others could join this thread and say if they agree or not with NT-PTMP being the 2nd most needed mode, would be great. Please, do not hesitate to comment. > > > Right now, I would not preclude the possibility that NT-PTMP support > might be added, but I could not give you a concrete time at which

CLI>realtime mysql status Connected to asterisk at 127.0.0.1, port 3306 with username askuser for 1 minutes, 34 seconds. Thank you very much for your kind attentino. You help is greatly appreciated. Thanks, Mark ------=_NextPart_000_05AD_01C88E72.41751FF0 Content-Type: text/html; charset="us-ascii" Content-Transfer-Encoding: quoted-printable <!DOCTYPE HTML PUBLIC

<br> ;<br> ; Display certain channel variables every time a channel-oriented<br> ; event is emitted:<br> ;<br> ;channelvars =3D var1,var2,var3<br> <br> So if you want fu_callerid, set:<br> <br> channelvars =3D fu_callerid<br> <br> And, once that variable is set, you should get a NewExten event, you<br> should see the following

, as it seems to be running Asterisk-11. =A0I&#39;ve previously installed A= sterisk-11+FreePBX in a VM, and this appears to be very similar. =A0Is ther= e any upside to using AsteriskNOW vs. Asterisk+FreePBX? Other than the obvi= ous fact that everything is nicely placed on an iso for ease of installatio= n?<br> <br> As for the actual upgrade, is it possible to step through each

<snip> Replaces: pickup-9582-c0a80101-d-4 at 192.168.101.102 <snip> This Replaces header refers to RFC3891 which is not yet supported in Asterisk (see http://www.voip-info.org/wiki/view/Asterisk+SLA) This INVITE fails with : <snip> chan_sip.c: Trying to pick up 7792 at subs <snip> app_directed_pickup.c: No target channel found for 7792. If I'm dialing *87792 instead

sk hangup after respectively 899s 894s 898s<br> <br> In logs I see<br> <br> WARNING[8213] chan_sip.c: Retransmission timeout reached on transmission 52= 2eec628683-uy8xshd6wc21 for seqno 102 (Critical Request) -- See <a href=3D"= https://wiki.a" target=3D"_blank">https://wiki.a</a><br> Packet timed out after 6401ms with no response (or

supported, beside Idle, On call and Ringing ? Can we expect this list to match DEVICE_STATE's one (UNKNOWN | NOT_INUSE | INUSE | BUSY | INVALID | UNAVAILABLE | RINGING | RINGINUSE | ONHOLD) > Might be worth seeing if other phones do the same. > > S > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by

supported by Asterisk for Video. I also find that video_caps branch has a fix for this problem, please can someone share more information about this and where i can find it ? I do not want those fmtp lines to be stripped. Suggestions to change the Asterisk config files, to achieve this are also welcome. Thank you. Best regards, Simith ------=_Part_17870_6007467.1218041254938 Content-Type: