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Couldn't see the forrest for all the fascinating tree-like applications that are out there: For future reference, see: http://www.voip-info.org/wiki-Asterisk+call+parking :-) -----Original Message----- From: Kris Boutilier [mailto:Kris.Boutilier@scrd.bc.ca] Sent: August 11, 2004 1:10 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Inband announcement of parking slot from app_parkandannounce? I'm trying to use Asterisk app_parkandannouce to build a global parking pool from within a couple of...

Hello all, I updated from CVS 3 days ago and now my IAX2 gateway is dropping calls without warning. It happens right in the middle of a conversation with no pattern. I never had this Problem before and am usually talking 2-3 hours a day. Is their a bug? Should I rollback? Cheers, Paul Seniuk -------------- next part -------------- A non-text attachment was scrubbed... Name: Paul

...is also an 'allowed' codec (w/5 licenses installed). During an average call 'iax2 show channels' provides: Peer Username ID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format 10.0.40.140 astpbx-woo 00002/00002 00005/00006 00040ms 0036ms 0000ms GSM Kris Boutilier Information Systems Coordinator Sunshine Coast Regional District

...modem traffic over the link (a credit card terminal). Obviously the modem audio isn't going to survive the G.729 codec process intact, so for the times the device is used I'd like to service calls from that device (and only that device) with a higher-data rate codec. Any suggestions? Kris Boutilier Information Systems Coordinator Sunshine Coast Regional District

...Handy for sites with departmental DID numbers who want to maintain their individual identities however it's not cheap. Telus's service also causes your corporate long distance phone bill to come seperated out and grouped based on the originating number - hence Station Level Billing. Kris Boutilier Information Services Coordinator Sunshine Coast Regional District

Unfortunatly no on both counts. The arrangement right now has: PSTN Trunks & Stations <-> Nortel Norstar#1 <-CT1-> Asterisk#1 <-IAX2-> Asterisk#2 <-CT1-> Nortel Nortstar#2 <-> Stations The Asterisk boxes provide Voicemail to their sites Norstars and intersite calls over IAX. Local Voicemail works flawlessly at each site but there have been reports of PSTN calls

...ess-routing' Aug 27 01:13:10 DEBUG[68621]: chan_iax2.c:2335 iax2_hangup: We're hanging up IAX2/astpbx-woodbay@astpbx-wharfrd/1 now... -- Hungup 'IAX2/astpbx-woodbay@astpbx-wharfrd/1' Bug, feature or other suggestions? Build is 'Asterisk CVS-HEAD-08/13/04-10:37:13' Kris Boutilier Information Systems Coordinator Sunshine Coast Regional District

...ebug the precise drop condition? I've Googled for more information on 'iax2 debug' but come up naught. > I've heard the jitter buffer is a bit buggy, have you tried > turning it off > completely? Is 'jitterbuffer=no' not sufficent to clobber that function? Kris Boutilier Information Systems Coordinator Sunshine Coast Regional District

Inquiring (management) minds want to know. I'm assuming it's because 'it's funny how simple it really is to write a really decent voicemail system'? Kris Boutilier Information Systems Coordinator Sunshine Coast Regional District

...Likely you can work around it by issuing an Answer() to the Norstar before dialing out. That will satisfy the Norstars ISDN timers. Also, try enabling extended results on the Norstar - you should see a 'recov time exp' error message displayed on the originating set. Hope that helps. Kris Boutilier Information Services Coordinator Sunshine Coast Regional District

I've got a need to do something like the following: [foo-context] exten => _.,1,SetCIDNum(123) exten => _.,2,SetCIDName(XYZ) include => local include => tollfree But of course, this example won't work. The goal here is this: if a call ends up being handled by the "local" or "tollfree" contexts, I want those SetCID*** commands executed. Otherwise, I

Hello, After solving the other "low hanging fruit" audio issues in our Asterisk PBX, we are left with occasional cases of severe echo which we have not found a solution for yet. Our system: - Asterisk 1.2.0-beta1 - TE110P on a PRI - TDM04 and TDM40, but these are unrelated to current echo issues - Fedora core 3 - Echo canceller KB1 Most calls have minimal, acceptable echo levels. But

Using TDM400's here and I have tried everything to cure the echo. I have used the Milliwatt test from the telco and from asterisk to tune RX/TX gain via a patched ztmonitor. What happens is I experience midcall echo. I turned on aggressive_suppressor and it seems to do great. The problem happens with misc. noise around the office will cause it to mute the other end of a phone call while

...ible. For the interested (or bored) far too much information about digit signalling can be gleaned from: http://ftp.tiaonline.org/tr-41/tr4138/Public/2004-02-Vancouver/TR41.3.8-04-0 2-002-L-(word)PN-3-4350.230RV3-Draft02-NetworkSignaling%20,DMcKinnon,AST.doc -----Original Message----- From: Kris Boutilier [mailto:Kris.Boutilier@scrd.bc.ca] Sent: May 28, 2004 2:50 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Problem with digits blending on inbound pulsed digits? I have a situation where I am receiving DID calls using Immediate Start Pulse signalling on a Loop Start trunk. The...

...an 'outside transfer' playback before the silence period. I have tried including an Answer() before the dial to patch the audio, but with no change. Obviously opening the channel to two-way audio before the dialing sequence is complete would be a security problem so, any suggestions? Kris Boutilier Information Systems Coordinator Sunshine Coast Regional District

...whereby they can hear the dialtone, digits and so on. Essentially bridge-before-connect, but with one way audio so they can't inject anything into the dial sequence. Can anyone suggest a method of achieving this from within the dialplan rather than modifying the Dial() application itself? Kris Boutilier Information Systems Coordinator Sunshine Coast Regional District

Before I relate the actual problem, some context. Callcentre environment, a few users testing a new digital dialer... 1. Agents are using Grandstream ATA HT486 and a small analogue dialpad with a headset. 2. SIP connection to Asterisk-1.2b1 3. IAX2 connection to ITSP provider. The call is initially set up in the following way. 1. Agent calls into a meetme conference room and subseqently stays

...l ID in the 'Display' element... However I understand from http://resource.intel.com/telecom/support/tnotes/tnbyos/2000/tn033.htm that there is no definitive standard for transmitting the name. So, should even I be expecting Ast to put the name on the wire when it's originating? Kris Boutilier Information Systems Coordinator Sunshine Coast Regional District

Hello I have a Grandstream GXP2000 with latest firmware. When I use it holding the handpiece I don't hear any echo - neither does other end. However, if I use it handsfree, the other end notices echo when they speak - ie their voice is echoy. I hear their voice being a bit echoy. Is this purely down to the IP Phone? Is there anything I can do about it? I considered buying a more

I am trying to work out IAX <--> IAX communications with my * box. I have a registration with iaxtel and I thought I would start there for my learning. I am able to call the number for Digium's support line (700-428-6000), but the sound is horribly chopping. Some reading revealed the jitterbuffer settings, so I enabled them in iax.conf. I have the following now: ; Inter-Asterisk