similar to: GSM gateways - which one ?

I was cleaning and working on laptops most of the day. Check my logs, I did plenty of work. -----Original Message----- From: "asterisk-users-request at lists.digium.com" <asterisk-users-request at lists.digium.com> To: "asterisk-users at lists.digium.com" <asterisk-users at lists.digium.com> Sent: 11/29/2008 1:13 PM Subject: asterisk-users Digest, Vol 52, Issue 81

Hi List; Is there an Digium cards support GSM SIM cards so we can fix an SIM card to be used for calls within mobiles as it is less rate? Or I have to use an FXS to SIM adaptor? If yes, then anyone advise a models and prices? Regards Bilal ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo!

Hi all, I just configured a quescom 400 to route all gsm incoming calls to asterisk, now i would route all outgoing asterisk calls to gsm port of the quescom. Anyone has any idea how implement it? I did a configuration but i always get this error -- Got SIP response 503 "Service Unavailable" back from <ip_add_quescom400> Thanks in advance. Giordano

Hi, I have a Portech MV-374 GSM Gateway and I'd like to send SMS from a web page to confirm the subscriptions. How can I achieve it? Is Asterisk of any use to send SMS with the Portech? I really have no idea because I know nothing about the whole SMS thing... Thanks, Niccol? -- http://www.linuxsystems.it

Hello, I am looking for a gsm gateway that is SIP based i.e no need of FXO/FXS analogue connection. I searched the email archives and found messages from 2008 but not sure how accurate these are. What do you use and how well it works ? The only sensible one I found is one made by portech and one that is made by Eurodesign. The one from portech is like a trunk while the one from eurodesign

I've an asterisk 1.2 connecting to a quescom gateway via SIP, the caller (asterisk) can hear the called, but the called hears nothing. Since both machines are on public ip, what other problem can it be ? There's one configuration working : lynksys pap -sip-> asterisk server -sip-> quescom this way both sides can hear voice but with : lynksys pap connected to a switch -sip->

More than 60% of our outbound calls are now to mobiles, so the time has come to whack in a gsm channel bank. Does anyone have any preference of bank ? Do you use a PRI or VOIP connection from the bank to asterisk ? Real-world experiences are sooooo much better than marketing blurb ;) We currently have a TE412P with a free socket, so we have a choice either way. I am looking for up to 30

I've an asterisk 1.2 connecting to a quescom gateway via SIP, the caller (asterisk) can hear the called, but the called hears nothing. Since both machines are on public ip, what other problem can it be ?

Hi, I am currently building a 1.4.4 Asterisk box for a client and they are interested in GSM functionality. Does anyone have any experience with a GSM card, preferably Quad Span (4 GSM modules or higher) for use in the UK. I have seen the Junghanns* version but I am not keen on the limitation of having to use a BriStuffed version of Asterisk. Do Digium make one ? as I am unable to find

Hello All * users. I have been looking for a way to allow GSM termination through Asterisk to occur. I am not a Telekom fundi and I have set up IAX2/IAX/SIP on asterisk with the ZAP channels via the Digium TDM 400P. I am unable to find any place that can tell me the cost of the VoiceBlue with a currency to I can calculate the cost of buying one. Alternativly - or just out of interist - I

We will be connecting our Asterisk server to ISDN 30 and intend using the Sangoma A101 card. The install location is in London (UK). Sangoma card at Voipon http://www.voipon.co.uk/sangoma-a101-pri-isdn-card-p-132.html?gclid=CI32vJz22I0CFQXklAodIgjHaA I would be grateful to hear if this is the right choice of card. Usage reports would be helpful. Regards Rory -- Rory Campbell-Lange

Has anyone been able to configure portech's mv-378 gateway with asterisk? I did the configuration as per the manual but it does not work. My server sees the portech gateway, but when the gateway is trying to register to my server it fails. It says peer is not suppose to register. The gateway and the asterisk box are on two different location (two network, 2 differrent IP address). I would

Hi All, What do you recommend? I was looking at: http://www.voipon.co.uk/sangoma-a200-fxo-fxs-analogue-card-pci-express-p-393.html But it will be 3 PCI slots. Thanks, Gavin.

Managed to get the portech 370 up and running with asterisk (even got the callerid working!), but was wondering how (if) it is possible to send / receive sms messages through the device . All I could find googling was people asking how ;( Does anyone have sms working with this device ? Julian ______________________________________________________________________ This email has been scanned

Hi List, I'm wondering if anybody on the list managed to get one of these beasts working with asterisk? FYI They're Windows NT embedded (yuk!) based H.323 / SIP compliant devices with a *very* complicated admin interface. Can't figure out how to get it working... yet :) Cheers, Jean-Michel.

Hi there, I'd like to know if some of you knew Quescom products (http://www.quescom.com) and if it is possible to make run Asterisk with a Quescom400. Thanks. -- Asega Adrole

I terminate some calls on a h323 device (a quescom gsmgateway) from asterisk 1.2.3 with ooh323, the customer is complayining about choppy sound on most of the calls, the only warning message I can see is : src/chan_h323.c:944 ooh323_indicate: Don't know how to indicate condition -1 on ooh323c_102 (the calls sounds perfectly with iax/zap termination and the quescom seems to work fine with

Hi, I am experiencing a weird issue with my MV-372. Mobile1 & Mobile2 are both registered to my asterisk server, I am able to use them for outgoing call with no problem, but when I call the sims in my gateway, they are routed to the right context/extension/priority, but as soon as I hangup, the sim unregistered from asterisk and tries to register with my the callerid of the last incoming

Hi, I have a problem with Asterisk-1.4.22 (with TB 2.6.2) & Portech MV-370, my problem is that when arrived an external call I don't view (on my internal phone) the phone number but I have the number extension that is configured on MV-370. The MV-370 configuration is: Mobile to Lan Table : 0 * 192.168.1.1 Lan to Mobile Table: 0 * # SIP Setting: Display Name: Portech User Name: 1005

This has to be an Asterisk based appliance no? http://www.truecall.co.uk/acatalog/trueCall_Features.html Looks pretty easy to setup using AstLinux or similar.