similar to: 1.6.1.10 Music On Hold

Hi everyone, I'm having an odd problem with one way RTP on SIP to SIP calls. I have two SIP servers, one is an Asterisk and the remote SIP server is a Nortel SIP server. When a call comes to the Nortel server through the PSTN and is routed to the Asterisk, audio is fine. Two way RTP and no problems. When a SIP client registered on the Nortel server calls the Asterisk, the Asterisk

Hello, When I bridge an incoming and outgoing call (attempting to simulate call-forwarding) I'm only getting one CDR -- that of the outgoing call. A (PSTN) calls B (residing on Asterisk) and the Asterisk calls C (cell phone on PSTN) and bridges the call. The only CDR created is from B to C. I have even tried using Answer() and ForkCDR() to get two CDRs, but to no avail. I am starting to

The Asterisk Development Team is pleased to announce the release of Asterisk 1.4.27, 1.6.0.18, and 1.6.1.10. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ These releases resolve a large assortment of issues reported by the community. For a summary of the changes in these releases, please see the release summaries:

The Asterisk Development Team is pleased to announce the release of Asterisk 1.4.27, 1.6.0.18, and 1.6.1.10. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ These releases resolve a large assortment of issues reported by the community. For a summary of the changes in these releases, please see the release summaries:

Hi guys, I am wondering (and have been unable to find out thus far) whether Asterisk sets some special channel variables or something when a call is transfered with the REFER method. Basically, I'm trying to figure out if it is possible to somehow get a transferred call back to the transferrer (as it is done with the built-in atxfer) after X seconds (or an unsuccessful attempt). Using a

Hi, I'm seeing weird behavior with AMI where no events are output until some input is detected (can be an empty line), at which time all the buffered output is spewed out at once. I am maintaining multiple Asterisk installations, and with one installation I have run into a weird buffering problem with AMI. The version is 1.6.1.11 in this particular case, which I am running at multiple

Hi, I'm having problem with call parking. When I park call, either via transfer to xten or park digit sequence from features.conf, I hear the parking lot number read to me and the user gets transferred. However, MOH stops for the caller the moment user is transferred. The user can be retrieved by dialing the parked extension and voice resumes. If the parked user hangs up, the channel state

Hi, Would you recommend some standalone SIP phones that work well with Asterisk? Personal experience preferred. Thanks, -- Raj

Hi, Just tried upgrading to 1.8 and ran into two problem immediately; 1. Caller-ID behavior is different -- now when I set the caller-id name to something with special characters (?, for example), the SIP INVITE now has %C3%96 instead of the ? character. I've tried doing Set(CALLERID(name-charset)=utf8) as well as iso8859-1, but it's always the same behavior. 2. My AMI scripts have

Using the Thumb-2 target we see that ORN ( a | ^b) and BIC (a & ^b) have similar patterns, as we would expect: defm t2BIC : T2I_bin_irs<"bic", BinOpFrag<(and node:$LHS, (not node: $RHS))>>; defm t2ORN : T2I_bin_irs<"orn", BinOpFrag<(or node:$LHS, (not node: $RHS))>>; Compiling the following three works as expected: %tmp1 = xor i32

On Jun 25, 2009, at 4:38 PM, David Goodwin wrote: > Using the Thumb-2 target we see that ORN ( a | ^b) and BIC (a & ^b) > have similar patterns, as we would expect: > > defm t2BIC : T2I_bin_irs<"bic", BinOpFrag<(and node:$LHS, (not node: > $RHS))>>; > defm t2ORN : T2I_bin_irs<"orn", BinOpFrag<(or node:$LHS, (not node: >

On Jun 25, 2009, at 6:06 PM, Evan Cheng wrote: > > On Jun 25, 2009, at 4:38 PM, David Goodwin wrote: > >> Using the Thumb-2 target we see that ORN ( a | ^b) and BIC (a & ^b) >> have similar patterns, as we would expect: >> >> defm t2BIC : T2I_bin_irs<"bic", BinOpFrag<(and node:$LHS, (not >> node:$RHS))>>; >> defm t2ORN :

Hi, Not sure where to submit this to so I'll try here. Below is the toneset for Iceland. Hopefully this can be added into the asterisk package. [is] description = Iceland ringcadence = 1000,4000 dial = 425 busy = 425/250,0/250 ring = 425/1000,0/5000 congestion = 425+250/250,0/250 callwaiting = 600/100,0/100,600/100,0/9000 record = 1400/500,0/15000 info = !950/330,!1400/330,!1800/330,0

Hello, What are the current methods for playing digits on different languages? I presume the big ones like German have been dealt with, saying 2 and 20 to announce 22. How is this currently decided? What about languages that say 20 and 2? Is there a way of configuring via config files or recordings somehow? Obviously you could record the sound file for 20 as "twenty and", but that

I have question thats not really about astrisk but I figure you guys are doing this sort of thing. We use Aastra 6757i phones. there is some support for XML. the question is how would i go about learning to customize these phones? _________________________________________________________________ Bing brings you maps, menus, and reviews organized in one place.

Hi Guys, Running Asterisk v1.4.26 (Elastix flavour) and have Aastra 9480i, 6757i, and 6730i, but none of them indicate the voic-email. Where should I look for trouble to find the root issue for MWI? Thanks, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100714/d268d1f8/attachment.htm

Hi, I've noticed that the UNIQUEID for a call is not the same in the Dialplan (when executed e.g. exten => s,n,NoOp(${UNIQUEID}) as it is when passed via STDIN to an AGI script. Is this normal, and is this supposed to behave this way? The UNIQUEID received in the AGI is usually .001 higher than the one in the dial plan -- but sometimes it is also a second behind. Here's an example

hello list i need your help please regarding an issue with snom300 and aastra6731i using asterisk 11.13.0 asterisk snom 300 8.7.3.25 astra 6731i 2.6.0.2019 i have configured the trunks like below 100 in snom300 200 in snom300 300 in aastra6731i 400 in x-lite the calls between x-lite and aastra ====ok inbound and outbound the calls between x-lite and snom300====> ok inbound and

I've got cups setup, to do the printwork. A setup, that was used with 2.2.3a of samba (same smb.conf), but it fails with 2.2.5. When using smbclient or the spoot client to connect to the samba printer resource, it tells me that it's a bad network name. Any known issues? -- --------- ?rn Einar Hansen email: orn.hansen@swipnet.se

The Asterisk Development Team has announced security releases for Asterisk as the following versions: * 1.2.36 * 1.4.26.3 * 1.6.0.17 * 1.6.1.9 These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of 1.2.36 resolves an issue where sending a REGISTER with a differing username in the From URI and Authorization header