similar to: IAX2/GSM VOIP troubleshooting

I am trying to work out IAX <--> IAX communications with my * box. I have a registration with iaxtel and I thought I would start there for my learning. I am able to call the number for Digium's support line (700-428-6000), but the sound is horribly chopping. Some reading revealed the jitterbuffer settings, so I enabled them in iax.conf. I have the following now: ; Inter-Asterisk

We have an application for Asterisk that will require connecting 144 fax ports into the system. Faxes will route externally over a PRI. The 144 ports are for local fax machines within the building. Not all will be faxing simultaneously. We just need to be able to provide ports in the building to plug in lots of fax machines. The plan is to run an Asterisk server for about 100 phones and these

Sorry if I post in this forum, this may be not the right one, but I hope to find in here some experts which could help me out. I have in one location 8 extensions from a Panasonic PBX ?KX-TD1232? connected to FXO Ports on an MultiVoIP Gateway from Multitech. On the other location I have 8 SP5100 IP-Phones from Micronet as remote extensions. Between the locations I have a VPN.

In the mailing list archives, I found a message that indicates that the IAXy has the ulaw, alaw, and g726 codecs, but I cannot find anything official on Digium's site about it. The Installation Manual has an example iax.conf file that indicates the ulaw codec, so I know that one is good. But we are thinking about using the IAXy over a VPN, to replace our MultiVoip. alaw and ulaw are

Anyone has configued Multivoip 3010 with Asterisk ? I am tring to configure this box to work with asterisk . Thanks Imthiyaz -------------------------------------------------------------------- mail2web - Check your email from the web at http://mail2web.com/ .

Hi all, I was wondering whether any of you have experience/info on Cable and/or ADSL modems that would come together with a SIP phone adaptor. What I am interested in is something that would plug directly into you ISP's cable (be it ethernet or adsl/phoneline), would combine a modem/router/nat such that on the other you could simply plug in your RJ-45 cable for your PC and a RJ-11 cable for

Hello Peter. Thank you for writing memdisk and isolinux. I've created a bootdisk with it, and it works gread with floppy images and noemulation (WinXP). I've created also a harddisk image with "nero Burning-Software". It works with the createted CD. But when I use the created harddisk image with memdisk. I'l receive error Message as follows. <<screen.jpg>>

Well I think we all need to look at something like this first. We will be one of the first people in Europe who will be selling this. If anyone is interested do drop me an email. Picture of the phone can be found here. http://cyber-telecom.net/wifi-gsm.jpg GSM / VoIP Over WiFi Dual-Mode Phone CYBER-TELECOM released the world first commercial GSM/VoIP Over WiFi dual-mode smart phone, in

Hi all, I spent most of the last weekend testing and trying to diagnose some mostly incoming call issues. I'll start with the easy one in the hopes it might have a positive impact on the others. First- I have an account with IAXTel. I can place calls to other IAXTel subscribers and also through IAXTel to landline toll free numbers and all works great. iax2 show registry shows I am

I created an account at IAXTEL.com to route 1-700-XXX-XXXX calls through. IAXTEL.com gave me a number (example) of 700-555-6226. I have made the following changes to my: /etc/asterisk/extensions.conf: [iaxtel700] exten => _81700XXXXXXX,1,Dial(IAX2/myusername:mypassword@iaxtel.com/${EXTEN:1}) exten => _81800NXXXXXX,1,Dial(IAX2/myusername:mypassword@iaxtel.com/${EXTEN:1})

I've been working with the old-style (pre-SIP) MultiTech MultiVOIPs, trying to make them work against chan_h323. With the voips in rfc2833 mode, Asterisk can detect DTMF fine from them, but when it sends DTMF to them, they lock up on the second digit, crying about an incoming fax. Has anyone encountered that problem (and found a solution)? Taking the other route, switching DTMF to inband on

Hello friends, I would like to thank you all for your advices. Rcmdr is gread. All the best, Mehdi ********************* Mehdi Zarrei, PhDPostdoctoral fellowUniversity of Toronto Dept. of Ecology & Evolutionary Biology Royal Ontario Museum Department of Natural History Green Plant Herbarium (TRT) 100 Queen's Park Toronto, On., M5S 2C6, CanadaE-mail:

Have you looked at his website, www.netenable.co.uk ? Looks like he pays bills the same way as he answers followups ;-) g -----Original Message----- From: asterisk-users-bounces@lists.digium.com on behalf of Peter Bowyer Sent: Sun 03-Dec-06 8:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] VoIP GSM Gateways Not very good at answering followups to

I bought a MV-372 for 2 SIM cards as the one channel model seems to work well (see http://www.voip-info.org/wiki/view/Setup+MV-370+GSM+Gateway+with+Asterisk). The setup is such: --------- Inet --> VoIP provider ---> POTS | | (iax2, NAT) | asterisk (on abox with iptables fw) | (SIP, LAN) |----------> SNOM190 phones | ----------> SIP-GSM-module ---> SIM

Been meaning to ask this for some time... no big deal, but curious. I have a single register statement in my iax.conf for iaxtel like: [general] port=5036 register => npi:mypassword@iaxtel.com <snip> However, when I restart *, I see: Registered to '69.73.19.178', who sees us as 205.221.193.101:5036 Registered to '69.73.19.178', who sees us as 205.221.193.101:4569

I'm looking at setting up a VoIP GSM gateway to connect to my asterisk box. What experience have people on this list have with GSM gateway hardware. I have been looking at the 2N voiceblue products. Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061028/87014bd9/attachment.htm

Hello, i've setup dhcp as such: yum install dhcp vim /etc/dhcpd.config # # DHCP Server Configuration file. #?? see /usr/share/doc/dhcp*/dhcpd.conf.sample? #ddns-update-style interim; ddns-update-style interim; ddns-rev-domainname "in-addr.arpa"; ignore client-updates; subnet 192.168.75.0 netmask 255.255.255.0 { ?????? option routers????????????????? 192.168.75.25; #Default

Hi all, I would like people to email me at 'leif at hacklocalhost dot com' some example configuration files for VoIP providers which * can register with. I am going to expand upon the FWD php "wizard" I created for these other providers, but I need some examples as I don't actually use anything but IAXtel and FWD. So far sipphone and iaxtel has been mentioned. I can

G'day list, I am getting a lot[1] of traffic on my Internet link, ICMP messages from 69.73.19.178 telling me UDP port 5036 is unreachable (this IP address belongs to iaxtel.org). I see from the wiki that IAXtel supports only IAX2 from December 2003. Fine, however it looks like my * still wants to try and register using IAX1, and I can't find how to turn this off. This situation is

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi! I live in Sweden and I am having problems getting asterisk to properly detect when a caller hangs up. And yes, I DO have disconnect-supervision on my line. Also asterisk sometimes misinterprets the disconnect-signal as another incoming call. This usually happens if I hang up first and then when the caller hangs up, asterisk treats it as a new