similar to: sip error logging

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi! Are hints not yet implemented in res_jabber? I have this here: exten => 3000,hint,gtalk/gtalk_account/mari.xxxxxxx at gmail.com But the hint doesn't show any difference. It always shows online on the phone and core show hints always shows that: 6003 at internal : SCCP/6003 State:Unavailable Watchers 0 6002 at internal :

Hi! My customer want's to allow calls to landlines in EU and US and disallow calls to cells in EU. Rest of countries are blocked. Country blocking is easy... Is there a service that allows checking phone number? Maybe some specific Enum? I ask for number and server responds with info, for example: "Cell Phone, Belgium" or "Land Line, Germany". -- Piotr Gorski

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, has anyone managed to establish an XMPP connection to the facebook Jabber servers? I'd like to send messages on missed calls vie FB. - -S - -- (o_ Stefan Gofferje | SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler & Koch - the original point and click interface -----BEGIN PGP

Is there any integration for ACT! and asterisk? I've googled for hours and haven't been able to find anything. Thanks David [cid:image001.png at 01CBE88E.66E8E450] -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110322/ecc019cc/attachment.htm> -------------- next part --------------

Hello, I have Asterisk-1.8.3.2, dahdi-linux-complete-2.4.1+2.4.1, and libpri-1.4.11.5 installed and running on a Ubuntu 10.04 server all built from source. Everything is working nicely except one small issue. The CDR records are stored in the CSV file correctly and complete. The MySQL storage is working as it should and is automatically updating all the fields except the CLID field. I have

Hi folks, I repeat "as is" the title of a post someone did a few months ago, since I am facing the same problem and did not see one single answer to his post. Maybe I'll be a little bit more lucky. When I'm trying to issue an Atxfer AMI command, in the asterisk 1.8 branch, what happens is that some DTMF's are sent, like this : [Mar 22 15:46:27] DTMF[5910]: channel.c:3900

I am becoming frustrated with our current VOIP provider. Does anyone have any suggestions for a provider that supports asterisk well and provides solid service? Voip-info.org has a husge list of providers, but it is impossible to tell the fly-by-night operations from the reputable providers. --Brent -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hey Guys! We have two Asterisk with A102D Sangoma cards now i want to connect them back-to-back over PRI line via Cross-cable so what would be the configuration specially timing source and all? anybody did it before like this ? I want to make sure everything before putting in production.. (saving my downtime) -S -------------- next part -------------- An HTML attachment was

I recently configured a SIP peer which i must specify my fromuser as my ten digit "DID". I send calls to this peer, but whenever Asterisk sends an options message, the fromuser is "asterisk". Is this a bug? Or is there some other config I must make ? register = 2155551941:123456 at 10.0.138.226/2155551941~600 [peer](!) type=peer context=inbound qualify=yes

Hi, I am having issues sending and receiving fax on my asterisk setup. Currently I have a server that has 2 x E1 TDM cards one is sangoma and the other one is openvox. Both support echo cancellation. One of the e1 is connected to our telco provider via mfcr2 where all our incoming calls originate. On the other end is a pri connection going to HICOM PABX where the local attached to a fax is

Hi, I am having a little problem and I hoped maybe I could get some help here. I deployed a Asterisk 1.8 server of my own to make SIP calls just between my friends. The server is configured with a public IP address. My friends and I are using "Acrobits Softphone for iPhone" as a client. I am using its push service which is hooked up to my Asterisk server. Now, the current situation is

I upgraded from a perfectly working 1.6.2 asterisk installation (including fax via app_fax_digium) to 1.8.0 this evening. All my custom modules (including swift <thanks darren!>) are working fine except for fax. When a caller connects, asterisk switches to the fax context and hangs up the call. i've captured with: core set verbose 10 core set debug 10 fax set debug on sip

After upgrading to Asterisk 1.8.0, I am finding that my outbound SIP calls are being destroyed after 1 minute and 20 seconds (80 seconds). Asterisk is sending a BYE message - I have no idea why. http://jeremy.kister.net/tmp/20101115/sip.txt for a sip debug. can anyone suggest how i can further deal with this? -- Jeremy Kister http://jeremy.kister.net./

The Asterisk Development Team has announced the release of Asterisk 1.8.2.3. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.2.3 resolves the following issue: * Reimplemented fax session reservation to reverse the ABI breakage introduced in r297486. (Reported by Jeremy Kister on the asterisk-users

The Asterisk Development Team has announced the release of Asterisk 1.8.2.3. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.2.3 resolves the following issue: * Reimplemented fax session reservation to reverse the ABI breakage introduced in r297486. (Reported by Jeremy Kister on the asterisk-users

I am trying to use SoftHangup in my dialplan, but it's either not working or I'm not using it correctly. when i'm on the console, i see: pbx1*CLI> core show channels Channel Location State Application(Data) SIP/vgw1-000000a2 2156181505 at inbound:1 Up AppDial((Outgoing Line)) SIP/143-0000009f s at macro-SaferSIPDial Up Dial(SIP/99302156181505 at vgw1,, 2 active

just thought this was cute enough to pass along, https://www.youtube.com/watch?feature=player_detailpage&v=GLwct15X_3g#t=135 -- Jeremy Kister http://jeremy.kister.net./

Running Asterisk 1.6.2.11 on debian 5.0.6 with mpg123 1.4.3 when i start asterisk, i immediately see two mpg123 processes spawned which sit there forever. I can't imagine it's normal behavior, but if it is, please explain :) # /etc/init.d/asterisk stop stopping asterisk. #[...] # /etc/init.d/asterisk start starting asterisk. # psg aster root 14573 1 0 16:29 pts/2 00:00:00

With app_swift 1.6.2 + asterisk 1.6.1.12, I've found that if you enter DTMF during cepstral playback, the first digit of ${SWIFT_DTMF} is [un]set in an odd way. for example consider: 999,1,Swift(some long message that you dont want to wait for|5000|5) 999,n,NoOp(DTMF: ${SWIFT_DTMF}) if while I am listening to the playback, i interrupt and dial: - "12345", SWIFT_DTMF is set to

somewhere along the way, i noticed incoming calls from google voice are no longer working on my asterisk 1.8.3.2 system. When the call comes in, asterisk immediately prints on the console: == Spawn extension (google-in, s, 2) exited non-zero on 'Gtalk/+12153930924-f947' [May 12 22:47:18] NOTICE[13835]: chan_gtalk.c:1977 gtalk_parser: Remote peer reported an error, trying to