similar to: Automated Call Testing - end-to-end - SIP Provider

We are using vpn routers to connect home users back to our office network. Basically, shipping a mikrotik router that 'calls home' and establishes a vpn connection for the pc and phone that are connected to the mikrotik... user plugs router in, plugs phone and computer into router, and that traffic is encapsulated back to our office... simple and straighforward. We would like to remove

We would like to present a toll free CallerID when making outbound toll calls. In the past, when our PRIs were directly connected to a Nortel CS1000 we could do this, without issue. Now that the PRIs are front ended by a mediagateway facing asterisk, we can no longer do this. Is it possible to set the billing number via a SIP header and set what should be presented as callerid as another header

Good day.. We have what I consider to be a large dialplan (-= 1501 extensions (2559 priorities) in 99 contexts. =-) If we have more than 10 or so channels up (all SIP, no TDM) and issue the "extensions reload" command.. quite often, asterisk will completely freeze up... requiring us to either kill and restart the process or restart the box... I should probably also share that when

a strange problem of multiple sip registrations and peer selection in sip.conf is calling for your suggestions!! let's examine this scenario: some numbers and passwords hidden with HHHs to protect the guilty :) I have 3 distinct sip subscriptions with cordiaip.net provider in US. For each of these i insert in sip.conf (with peer name differences and relefant number/password differences,

Hi, Anyone know any decent trunk provider in Canada that can offer multiple channel trunks (16channels) via IAX or Sip trunking? Having some pleasant experience with IAXTEL out of Denver, though they don't offer services into Canada. Please let me know S.

Ladies, Gentlemen We are experiencing an unusual problem in our asterisk 1.4.34.. We are attempting to determine if channels are in use before paging to them. This works correctly, as in it pages the phone.. however, we see the error message below on the console... after googling, we discovered limited information regarding the issue... -- Executing [NPANXX7298 at from-pstn:1]

Greetings!! We're getting some strange results using call files.. no matter the technology, DAHDI, SIP, etc., we get a "Call failed to go through, reason (3) Remote end Ringing" message when attempting to originate a call from a call file. Numbers changed to protect the innocent.... using call file.... //------------CALL FILE------------// Channel: DAHDI/g1/918005551212

Is there any built in functionality when using Originate to retry a call based on the DIALSTATUS? Similar to the .call file where you can set max retries and time between them? I've tried putting the logic in an outbound context/macro, but it just times out if the time between retries is too long...I suppose your not supposed to leave calls in limbo too long. Is there any other method

Hi all, I have a budget to work with and was wondering if there are any folks providing SIP/IAX2 trunking for unlimited inbound/outbound for a flat rate? We're in the budget range of roughly $5,000 a month and we need multiple channels per DID. I'm not sure if something like this is feasible in the world of VoIP -- and I only need to be able to make domestic/USA calls. Thanks for any

I need SIP trunking from a high quality business service provider for 25,000 SIP minutes growing at approximately 10% each month. Currently we are using exgn.net to provide inbound 800 (Costs $200 for approx 10,000 minutes a month) and we are using broadvoice.com for outgoing calls (Costs $100 for approx 15,000 minutes a month. We have two unlimited business plans with them.) The reason I want

Can someone help me use Asterisk to provide SIP trunking to another PBX? I'd like to insert an Asterisk platform between a network PRI and an existing PBX. Incoming calls would (based on DNIS in a SQL table) go to PBX "A" on a PRI channel or PBX "B" as SIP. Outgoing calls from either PBX would select an available network PRI channel. --Don Don Kelly CT Magic 612

>Our user places a call, the gateway responds with no sound at all, or >hangs up, or gives busy tone. > >How can we get to the next provider? > >I have now: >exten => _9011Z.,103,Dial(SIP/011${EXTEN:4}@provider-a) >;exten => _9011Z.,103,Dial(SIP/011${EXTEN:4}@provider-b) >;exten => _9011Z.,103,Dial(SIP/011${EXTEN:4}@provider-c) >exten =>

In the case of Search::Xapian 1.2.x and xapian-core 1.4.y, $msg is deliberately unset because the two versions should be compatible. Since this is intended to be an escape from this conditional, it should be respected by the AUTOMATED_TESTING check, so automated testing with compatible libraries doesn't nevertheless result in no Makefile. --- Makefile.PL | 2 +- 1 file changed, 1

Hello friends, Any advice on automated testing of a web application with Prototype-based Ajax stuff? We were looking at something like Actiwate (Junit/HttpUnit stuff), but they don''t support XMLHttpRequest (yet). Using the new JS unit testing I''ve seen here doesn''t look like it would work, as we''re talking about multi-page as test vectors: 1. go to

Is there a way to automate testing of fxruby applications ... something like Jemmy for Java Swing? -- R. Mark Volkmann Object Computing, Inc.

On Sun, Sep 15, 2019 at 05:29:02PM +0000, Alyssa Ross wrote: > In the case of Search::Xapian 1.2.x and xapian-core 1.4.y, $msg is > deliberately unset because the two versions should be > compatible. Since this is intended to be an escape from this > conditional, it should be respected by the AUTOMATED_TESTING check, so > automated testing with compatible libraries doesn't

Hi all, I have a small system of two hardware boxes (residential gateways) running Linux with Asterisk on them. Each RG has some FXS ports to which analog telephones can be connected. I already had a working system including an external SIP provider, where both RGs would register to that provider with a telephone number and they could call each other via that telephone number. Each RG had a line

> Thanks for spotting, applied in git. Where is the appropriate git repository? I looked but couldn't find it, so ended up just having to make my patch from the release. -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 832 bytes Desc: not available URL:

On Thu, Sep 19, 2019 at 08:12:58PM +0000, Alyssa Ross wrote: > > Thanks for spotting, applied in git. > > Where is the appropriate git repository? Sorry, it's a bit obscure: https://git.xapian.org/?p=xapian;a=tree;f=search-xapian;hb=refs/heads/svn/1.2 We've now retired the 1.2 release branch apart from this directory so it perhaps should be re-homed in its own repo, but

>> > Thanks for spotting, applied in git. >> >> Where is the appropriate git repository? > > Sorry, it's a bit obscure: > > https://git.xapian.org/?p=xapian;a=tree;f=search-xapian;hb=refs/heads/svn/1.2 I don't see my patch applied there -- what am I missing? > We've now retired the 1.2 release branch apart from this directory so > it perhaps