similar to: Handoff dial control to dialplan after AMI Originate

Hello, I have asterisk 1.2.9 running on a Debian sarge server, my outbound dial plan looks something like this: [outbound-longdistance] exten => _91NXXNXXXXXX,1,Dial(${OUTBOUND1}/${EXTEN:1}) About every other outbound call we make, we get the 'you must first dial a 1' message from our phone provider. It only seems to happen every other try if we try to make multiple out bound

[Jul 2 09:31:16] VERBOSE[2682] logger.c: == Refreshing DNS lookups. [Jul 2 09:31:16] NOTICE[2682] dnsmgr.c: host 'outbound1.vitelity.net' changed from 64.2.142.17 to 64.2.142.29 [Jul 2 09:31:23] DEBUG[2711] jitterbuf.c: Attempting to exceed Jitterbuf max 600 timeslots And the calls are dropped. I fixed this by turning off enable in dnsmgr.conf My question is: Do you attempt to

I'm using ezstream-0.1.2 KJ -----Oorspronkelijk bericht----- Van: Joel Ebel [mailto:jbebel@ncsu.edu] Verzonden: donderdag 3 februari 2005 21:05 Aan: Klaas Jan Wierenga Onderwerp: Re: [Icecast] Stream drops during handoff. Suggestions? Thanks. I'll have to try that. I wonder why ezstream would ever stop sending data for that long though. What version of ezstream are you running? Joel

How do I do this 1) ZAP-> * -> IAX(1) ------> IAX(2) -----> DG104S ------> Handset 2) No Answer on Handset 3) Back to IAX(1) 4) IAX(1) tries a cell phone 5) Still no Answer 6) Local * Voicemail. I have 1 working, and I had 4 working when there was only one box, i.e. when the handset did not answer the DG, asterisk went to the next step. Now that I have step 1 going to another

I have a hardware FXO/FXS which handle my voip calls, and they support G723 internally. Asterisk hands off these calls just fine, and everything works, as long as I don't want PBX menues available... The problem is, once I want it to return messages, it will only return them as GSM... which is fine, since my FXO/FXS support multiple coders. However, even though Asterisk lets me specify a

So, when I get no comments on this at all, either here or on any of the forums, does that mean nobody knows what I'm talking about?? Or does nobody know the answer?? Or is it just a stupid question and nobody wants to bother telling me where to look?? It *is* a question that I have to answer somehow; I've read all through TFOT and see nothing relevant to this issue. It's silly to

Sorry if this has been asked before, but I've searched high and low for the last couple days for an answer. An Internet radio station I DJ for is using Shoutcast and MP3, but we are considering moving to an Icecast/Ogg Vorbis combination instead. We work in 3-hour shifts. When we hand off, the DJ on-stream stops teh encoder, shouts "go" on IRC, and the DJ in line starts his

Hey All, I am setting up a network of Asterisk servers using IAX2. I am wondering if it is possible to disable the handoff feature? At the moment I have 4 asterisk machines, 3 are at SOHO offices and 1 is centrally hosted in a data centre. In addition the central machine has an IAX2 link to a VOIP provider (and might be set up with more in the future). All calls are routed through that

let vga16fb claim 0xA0000+0x10000 region as its aperture; drm drivers don't use it, so we have to detect it and kick vga16fb manually - but only if drm is driving the primary card Signed-off-by: Marcin Slusarz <marcin.slusarz at gmail.com> Cc: James Simmons <jsimmons at infradead.org> Cc: Dave Airlie <airlied at redhat.com> Cc: Ben Skeggs <bskeggs at redhat.com> --- no

git://git.zytor.com/users/genec/syslinux.git Branch handoff-mbr-for-hpa This is a piece of code that can be used in place of a MBR or VBR/PBR (master boot record; volume/partition) in order to examine the data handed to the respective boot code. AX, SS, and SP are destroyed before examining anything. I set an internal restriction that limits it to 420 bytes such that it could be used with a

Hello, We have a Kamailio SIP Proxy in front of our Asterisk cluster for incoming calls from our carrier. The sip.conf looks like this: [kamailio1] type=friend host=10.0.0.1 context=incoming disallow=all allow=alaw All calls hit the incoming extension. In the extensions.conf we have multiple extensions configured, but now I have to add one which uses the special h extension to perform a CURL

Hey Folks, Sorry if this has been asked before; hard to get a usable search on 'isolinux OR syslinux pxe-boot' with just the info I need! I work in a large organization, and the group which manages the network may not be the guys managing the hosts. I have a network and dhcp server, and pxelinux is working very well on that net. I want to make my net-install resources available to

Hello, Does anyone know how frequent segment faults occur in the current LTS release (version 11) and in the future LTS release (version 13)? We are currently using 1.6, which frequently throws unexplained segment faults, that's why we are considering to upgrade to the latest LTS version. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm using Asterisk to bridge the incoming call to another destination using the Dial command. However, when an anonymous call comes in then privacy information is not passed into the B Leg. For instance, the Privacy header and P-Asserted-Identity aren't copied to the B Leg. Is there an option to give to the Dial command, or another variable to set, to make Asterisk copy such

It removes a hack from nouveau code which had to detect which region to pass to kick vesafb/efifb. Signed-off-by: Marcin Slusarz <marcin.slusarz at gmail.com> Cc: Eric Anholt <eric at anholt.net> Cc: Ben Skeggs <bskeggs at redhat.com> Cc: Thomas Hellstrom <thellstrom at vmware.com> Cc: Dave Airlie <airlied at redhat.com> Cc: Peter Jones <pjones at redhat.com>

Hello, I'm researching the possibilities of multiple communication platforms like Asterisk and FreeSwitch for handling a dynamic sequence of applications to execute, like Playback, Read, etc. This only applies to originating a call from an external application by using the AMI Manager and the Originate action. I need to know the following: 1) Does the Originate action support multiple

It simplifies nouveau code by removal of detection which region to pass to kick vesafb/efifb. Signed-off-by: Marcin Slusarz <marcin.slusarz at gmail.com> Cc: Eric Anholt <eric at anholt.net> Cc: Ben Skeggs <bskeggs at redhat.com> Cc: Thomas Hellstrom <thellstrom at vmware.com> Cc: Dave Airlie <airlied at redhat.com> Cc: Peter Jones <pjones at redhat.com> Cc:

Hello, I'm trying to execute a stored procedure on a MSSQL Server from the dial plan, but it's not working. I'm getting the following error: Unable to execute query.... Asterisk has been compiled with UnixODBC, and I've done the necessary configurations in func_odbc, res_odbc and odbc.ini. Has anyone done this before with success? -------------- next part -------------- An HTML

Hello, I'm noticing strange behavior in one of our Asterisk nodes where the ACK is always sent to the proxy, but RR is not enabled for calls. The proxy drops the ACK. I'm using the AMI interface to originate a call: Action: login Username: myusername Secret: mypassword Events: on Action: Originate Channel: SIP/<SOMENUMBER>@proxy1 CallerID: <SOMENUMBER> Application: Playback

Hello, I'm new to Echo Cancellation and I was wondering how it is handled/works on pure VoIP networks using Asterisk? I did some research on the internet about EC on VoIP networks, but I can't really put a grasp on it. We currently have some Echo Cancellation chips on our Digium cards, but are planning to move to a full VoIP network based on Asterisk. So no more ISDN in the voice path.