similar to: Queue with cell phones

While most cell phones are fine, some cell phones don't seem to produce DTMF digits that Asterisk/Zapata will detect. One of our salespeople has an AT&T model that never gets any digits through. Is there a known solution? I understand, of course, that with cell phones the DTMF tones are actually created at the base, and that pressing the key longer usually does not result in longer

Greetings, How can I call cheap to UK cell phones. I am located in Toronto, Canada, but need to call UK cell phones both from Toronto and London. -- Zeeshan A Zakaria -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080226/480f7a28/attachment.htm

Hello, all. Our first major asterisk system is just about ready for production. However, we noticed that our outbound SIP callers did not receive rings when dialing cell phones. Land lines were fine. We "fixed" this by setting progressinband=no in sip.conf. However, I gather this places extra load on the asterisk server and I'm not sure that it always conveys accurate status

Hey guys, I just installed SCDA, and also installed directx9 with winetricks if it is needed for the game. I have a laptop (ASUS F5SL AP104) with the following setup: Intel Core 2 Duo T5750 3 GB RAM AMD/ATI Mobility Radeon HD3470 (worst VGA ever used, but don't have better though) Ubuntu 8.10 Intrepid Ibex, wine-1.1.15, latest ati drivers Here are the terminal outputs: First with original

I have a very strange problem I'm hoping someone has encountered already. I've scoured the internet for an answer to this one. My phone company provides no disconnect supervision. Hence I'm forced to use the busydetect feature. I have a TDM400 with two FXO ports. If I call from an internal extension to a landline and then hangup the landline Asterisk detects the busy signal

Hello, MOH always is choppy when someone calls from a cell phone to my pots or nufone 866. It sounds fine when it originates from a land line. I use zaptel hardware, and plenty of resources. I have tried to use different songs. None have the id3 tags, I tried the custom settings with -q -r 8000 -f 8192 -b 2048 --mono -s. Tried permanent resampling to 8khz, 16bit, filterd with lame -q1. I

I have two different inbound DID providers and on each network I have had reports in the last several days from callers that when they try to call me from a cell phone their provider reports that they are trying to make an international call and thus doesn't let them complete the calls. In each case these have been local calls for them. Has anyone else had this experience? MARK.

Hello, My company has a 800 number that we use for customer service. However, instead of having our customer service reps at the office, we route these calls to their cell phones using a service provided by gosolo.com. However our current system isn't ideal because it will call each number in order causing longer and longer waits when we've got people on the line. Our goal is to set

My settings for the queue.log are in the [general] section of logger.conf I'm running 13, I didn't see what version you said you were running. If I wanted to add a LOCAL channel to my queue I'd do it as member => LOCAL/7124 at kiniston-intern,0,John,hint:7124 at kiniston-intern On Thu, Nov 15, 2018 at 2:38 PM Ivan Demkovitch <idemkovitch at yahoo.com> wrote: > John,

Hello, is anyone on the list using a normal cell/mobile phone which is able to act as a SIP client over WLAN? Or has anyone heard of a SIP client for cell/mobile phones running windows mobile 6.x? The phone should use SIP, when the asterisk server is reachable and should automatically switch to a German telco if it is not reachable. Thanks for any hints, Stefan --

Hi, I've a DID number that gets passed to three internal phones and a cell phone via my outbound IAX trunk. If the cell phone is off or out of coverage, its voice mail captures the call. What's the best way to avoid this? Is there a recommended way to force the cell phone user to press 1 before the call is passed there ala google voice? Or is there another way to detect the presence of

I'm having a big problem where digits dialed from certain cell phones are too short to be recognized by my asterisk server. I'm running AAH 2.8. Some cell phones don't allow the caller to hold down the digits and have the tones play as long as they hold them down for. They just play a short tone no matter how long you hold down the digits for. Has anyone run into this before, and if

I have an Asterisk 1.4.2 system installed at our office, and have a few 'on the road' sales people that want to make calls from their cell phones in transit, but they are complaining that people returning calls that they make from their cell phones are simply just using the CID that is coming from the cell phone which is causing them to get phone calls outside of business hours. What

Hello, I have queues.conf setup with a group like so: [Sales](StandardQueue) announce = first member => SIP/FF4C119EEBF8-SLS member => SIP/FF9EF375CCFC-SLS member => SIP/13145555555 at callcentric ;Eric's cell member => SIP/FF1565AABB2D-SLS ;Eric's Yealink So, my idea here that it should ring all 4 phones at the same time. And it does work but randomly.I did trace a call and

Hello All, I apologize if this question has already been answered but how do you transfer a call to a cell phone or another land line outside the PBX? Setup I have two pots lines into my current Asterisk Box. I have an outsides sales guy who wants to work off his cell phone or transfer his calls from his extension and the main sales extensions. How can I do this right? -- Otis

I don't really know the name of what I want to look for but maybe someone could tell me if it would be available. I have a number of old analogue cell phones laying about here and I was thinking it would be useful if I could set up a short range base station for them that would cover maybe an acre or so. What I would like to be able to do is use it to connect into Asterisk and this way

I'd like to integrate a few peoples cell phones into our asterisk system as we're walking around a data center some days and carrying another cordless phone just doesn't seem to make sense. Forwarding is easy, however once forwarded I want to be able to flash and transfer back to another party / voice mail / etc. How can this be done? As far as I know there's no way to generate

Yes. http://lists.digium.com/pipermail/asterisk-users/2005-February/087538.ht ml -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Nitesh Divecha Sent: Wednesday, March 02, 2005 2:56 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Call Forwarding to Cell Phone,

Currently the plan is to forward all PSTN calls on our 2 incoming PSTN lines & 2 remote toll free's via IAX2 to staff. 3 different delivery methods 1) Users local to the office where the lines come in with GS/PSTN phones 2) IAX2 to remote location * server then Cisco 7960 on that lan 3) IAX2 to remote softphone That assumes they are in the office though. Recently I've been out more

We have set up an HP DL380 with 3 4BRI cards, Fedora core 2 (kernel 2.6.10) and asterisk (bristuff-0.2.0-RC7f with asterisk 1.0.5). 4 ports are configured in TE mode and connected to the PSTN; the other 8 are in NT mode and connected to isdn phones. the other outbound calls to PSTN are fine, however, when we call cellular phones, often audio is one-way (i.e.: the cell phone user can not hear,