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Is there an advantage to having a Queue members URI in the form: SIP/User (or indeed IAX2/User) Over Local/<number>@context ? I know that the latter will allow you to do things like set counting logic etc. through dialplan operations, but the former appears to be a more direct route to calling the party. (and if need be, there is the ability in queues to run a script on connection iirc).

Does anyone know why when calling out with an ATCOM AG-188 registered with IAX (haven't tried SIP), there is no ring. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070222/a4f29a97/attachment.htm

On Mon, 2007-04-02 at 16:30 -0700, asterisk-users-request@lists.digium.com wrote: > Date: Mon, 02 Apr 2007 20:26:09 +0100 > From: Thomas Kenyon <digium@sanguinarius.co.uk> > Subject: [asterisk-users] Re: On Topic: Cheapest Asterisk USB Key? > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <461158D1.2020408@sanguinarius.co.uk> > Content-Type: text/plain; charset...

I know I must be being daft, but is there a way to set which context the queuing system uses when it dials the operators/agents? By default it appears to use the default context. I've looked through voip-info.org and can't find anything, someone please put me out of my misery.

I have an asterisk 1.2.9.1 machine with zaptel 1.2.6 running. On the TDM400P, I have 1 FXS port and 3 FXO ports. dmesg reveals: Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.2.6 Echo Canceller: KB1 PCI: Found IRQ 10 for device 01:01.0 PCI: Sharing IRQ 10 with 01:05.0 Freshmaker version: 73 Freshmaker passed register test Module 0: Installed -- AUTO FXS/DPO Module 1:

I have been seeing this problem for a long time and it occurs in 1.4.0b2 (as well as 1.2.0-1.2.12.1). If the internet connection is lost and I have SIP services that require me to register, any SIP devices attached to the system stop working. I have an IAX phone connected to one of my servers that I've been having this problem with which will work fine (and filover to the PSTN) the

I'm finally migrating from the 6/7/05 CVS version to 1.4. Had quite a run, I have to admit. Asterisk itself only segfaulted once or twice, but the dns issues have been bothering me. And the box just needs to go. Everything is going on a Ubuntu 6.06TLS server, that's been perfectly stable. I had 1.4.1 installed and running, but not configured. Yesterday I upgraded to 1.4.11,

The Asterisk and Zaptel development teams have released Asterisk 1.4.1. This release contains a very large number of bug fixes, including a fix for the recently discovered security vulnerability. It also contains a complete rewrite of the Shared Line Appearance (SLA) support that was first released as part of Asterisk 1.4.0. The new version of this functionality has been tested against a variety

The Asterisk and Zaptel development teams have released Asterisk 1.4.1. This release contains a very large number of bug fixes, including a fix for the recently discovered security vulnerability. It also contains a complete rewrite of the Shared Line Appearance (SLA) support that was first released as part of Asterisk 1.4.0. The new version of this functionality has been tested against a variety

Hi, I`m trying to use Chanspy for a customer that wants to listen to his employees so he can train them better (or so he claims). In any case, it looks simple but there is something I`m not doing right. When a call is incoming, I set SPYGROUP using Set(SPYGROUP=1234) When I use, on another phone, Chanspy(|qg(1234)) Which should allow me to listen to conversations that hit the first (Set

...olaris users out there dont have much > support when it comes to G729 codecs, a real pity really, this stops > some large scale roll-outs. > > > > ------------------------------ > > Message: 20 > Date: Tue, 15 Jan 2008 09:05:35 +0000 > From: Thomas Kenyon <digium at sanguinarius.co.uk> > Subject: Re: [asterisk-users] G.729 pre-compiled binaries and Asterisk > 1.2.x. > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: <478C775F.4010002 at sanguinarius.co.uk> > Content-Type: text/...

Hello. I am currently testing an asterisk server using the default codecs, I have allow=all, and noticed everytime I test it in a wireless lan the latency rockets off the roof to over 1000ms. I would like to test g729 since it uses less bandwidth but, read somewhere I have to buy a license per every channel I have. Does this means if I have my server connected with 10 sip clients I need to buy a

Friends, Thanks for all the feedback. If you have additional success stories or important issues, feel free to continue the discussion. I've learned a lot from your input. As a developer, I spend too much time in the bug tracker, working with particular bugs, so I often wonder how on earth anyone can use this buggy platform for anything business-like. It really feels good to get

I run Asterisk 1.2.7 (upgraded from 1.0.x to 1.2.x etc.) I notice that if I run the Management, the banner message I get is: Asterisk Call Manager/1.0 According to voip-info I should be getting Asterisk Call Manager/1.2 Is this difference significant? I also get this with a machine that is in active use running 1.2.7. I can't see anything referring to it in manager.conf. More

I have (more than 1) provider that I receive calls from using IAX, and I have 2 IAX deskphones, all work fine except for some reason with 1 provider, when the call comes in, it doesn't match up with the incomingcall context. (A bit worrying, since I don't want people to be able to relay calls off me.) in iax.conf I have: [ipcomms] type=user nat=yes dtmfmode=rfc2833 host=71.16.179.149

Using Asterisk 1.2.12.1. I have 4 queues running on a server with various handsets logged into them. When a call comes in, asterisk tries forwarding the call to all handsets, including ones that are in use (whereby it gets a BUSY HERE response, which is all what you'd expect after all asterisk doesn't know how many handsets are on each channel). If all the handsets are in use, then

Periodically (as in sometimes several times a day and sometimes never) I get A channel.c: voided initial deadlock for '0x82*****', 10 retries! The ***** figure is different each time. When this happens an active call (in or out) is dropped. The setup is as follows: handset --SIP--> Asterisk 1.2.12.1 --IAX2--> Terminating supplier (don't know which software they are running).

I haven't tracked this down to anything on my system yet, but has anybody else upgraded to 1.4.0b4 (from 1.4.0b2) and found that asterisk core-dumps on startup? The last few lines in messages before dump are: [Dec 16 10:44:03] WARNING[7958] translate.c: plc_samples 160 format 6 [Dec 16 10:44:03] NOTICE[7958] chan_agent.c: No agent configuration found -- agent support disabled [Dec 16

Has anyone else had any difficulty with calls Originating from the PSTN being passed to asterisk 1.4.0 unsing a linksys SPA-3000? I've not had enough time to track down what's happening but with 1.4.0, When a call comes in, it is passed to asterisk and then forwarded to the extension that rings, but when the extension is lifted the call hangs up. This does not happen with 1.4.0b2

I have an ITSP provider that will only deliver calls using SIP registrations (would prefer delivery to static IAX or SIP url, but hey), periodically their servers don't respond to a renew request, and when this happens the sip stack in asterisk (1.4.0) stops working until either a SIP reload is issued (or sometimes a restart now). I'm wondering if this can be solved by installing