similar to: How To - Building a VoIP-PSTN Gateway with Asterisk

In trying to get the asterisk and faxing working I had to resolve to using iaxmodem and hylafax. I have incoming working, but outgoing the other fax rings but it would appear from web searches that the fax signals are too low to be "heard" I can read about rxgain and txgain for zapata. my fax setup goes direct from aterisk <-SIP-> SIP Provider <- Fax Machine -> It never

<P>Hi List,<BR>is it possible to connect an existing telefon system with a stanard AVM ISDN card to the telco?</P> <P>&nbsp;</P> <P>[ISDN from Telecom supplier] ---- [Aterisk Box] ----- [ existing telephon installation]</P> <P>&nbsp;</P> <P>kind regards,<BR>Patrick</P>

> On 2020-06-06 10:38, Antony Stone wrote: > On Saturday 06 June 2020 at 09:18:11, Fourhundred Thecat wrote: > >> In a situation when I start asterisk, and the remote database is >> unreachable, asterisk waits for several minutes before it actually >> starts (before it loads sip module, etc). >> >> And when database is unreachable during operation, when call

Hi Asterisk-users, I am working with Aterisk Manager API's. I can login successfuly with the following. char buff[256]; strcpy(buff, "Action: Login\r\nUsername: admin\r\nSecret: unix\r\n\r\n"); send(msock, buff, 255); Now I want to try Action: Originate, therefore I tried the following char buff1[256]; strcpy(buff1, "Action: Originate\r\nChannel:

Hi.. I just wondering why DTMF are not recognized by aterisk on incoming calls from my SIP provider ... ANy suggesteions ?` /Mike

Hello, I'm very new with * and I would really appreciate some help to implement a SIP to PSTN Gateway. My current scenario includes an * box with a TE405P board. I have a 1.5Mb connection to the outside world (using a router with firewall capabilities) and channel banks that allow me to connect the T1s coming out of the TE405 board to PSTN network (carrier). I need to configure * to

I'm looking for recommendations for UK-based VoIP-PSTN gateways. They should ideally offer: - IAX connection - Multiple simultaneous calls on a single account - Lower call rates than BT Business - Auto-top up or invoicing in arrears I can find several that offer one of these facilities, but none that offer all. Thanks! -- David Gurr Congruity Ltd. Hemel Hempstead, UK

Greetings, We are currently testing a Sipura SPA-3000 as a gateway from our Asterisk system to a PSTN line for 911 access. We have a number of locations and want to place an SPA-3000 in each, connected to a PSTN line that will provide the correct ANI/ALI information to 911 for each location. It all works great, except for a reasonably significant (4 seconds) delay between when the SPA-3000

Hello, I'm very new with * and I would really appreciate some help to implement a SIP to PSTN Gateway. My current scenario includes an * box with a TE405P board. I have a 1.5Mb connection to the outside world (using a router with firewall capabilities) and channel banks that allow me to connect the T1s coming out of the TE405 board to the PSTN network (carrier). I need to configure * to

Hi, I just setup two voicemail boxes - but after they record the wav file, they don't seem to email it even though i have the correct format in my voicemail.conf file. Aterisk doesn't give me any feedback besides the recording of the voip file... and then stays there. Secondly, I can't get the Playback() function to work. Thanks, Imran

Hi there The "dnsmgr" in Aterisk 1.4.0 seems not to work. I enabled "DNS lookups" in dnsmgr.conf but after reloading the conf files * never refreshes DNS lookups. Any ideas how to debug this issue? Thanks in advance Housi Mueller --------------------------------- Don't pick lemons. See all the new 2007 cars at Yahoo! Autos. -------------- next part

Hi folks, I'm experimenting with Heartbeat and whilst I have it running in an active/standby configuration I cannot get Asterisk to perform properly. I'm able to start the asterisk software (I imported the aterisk start file from /etc/init.d into /etc/ha.d/resource.d) with the heartbeat software but heartbeat continues to start more Asterisk instances. Any ideas? Mark

Hi all, I have situation in that we are renting out a room in our office building to another company, including the provision of phone lines via our Aterisk box. To keep billing simple, we would like them to be billed separately for all their calls. We think we can achieve this by them leasing their own outbound IAX trunk (which our providers have said is no problem) and having all their

You can even set it to zero. Mine works well when in zero. The line pick up immediately :> -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Chris Stenton Sent: Thursday, February 09, 2006 6:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SPA-3000 VOIP-PSTN

As a newcomer to Asterisk, you will not be welcomed with open arms. First, you will find almost no documentation on it's features. Second, if you try to ask questions, you will be flamed and pointed to worthless how-tos and 'the wiki'. These worthless documents can only be useful for explaining how things work to those already in-the-know. Lastly, Asterisk is so bug ridden, expect

Can anyone recommend a good voip provider? Thanks

OK, I think I may have found the problem for myself at least. Actually, a friend of mine suggested it. Apparently, Asterisk is a little too fast for the card. Placing a "w" in front of the number to insert a pause looks like it did the trick! Dial(ZAP/1/w5555555) Looks like it gives the card a chance to come online? So, at least in this case, it was not that Asterisk was keeping

Sure would like to hear experiences using various FXO to VoIP gateways with *. It seems that any thread that has anything to do with problematic FXO interfaces goes on forever with speculation about everything under the sun. Unless there is someone out there with the engineering experience to build a better one it is a waste of time, let Digium deal with it. If the TDM400P can ever be made 99.99%

We manage our own VOIP solution using Asterisk. Has anyone had success with an IP-PSTN provider? I'm looking for someone to terminate SIP calls to the PSTN in the Seattle, Washington area. (vice-versa as well if possible) Yes, I could do it myself via asterisk and digium cards but I would like to consider other options. Any opinions? Thanks, Chad -------------- next part

Now that I have most everything actually working I've noticed that about every 3-4 days on average..... and at worse... Once a day my asterisk box seems to lose it's registered state with our sip provider and no longer will take any incoming calls. The caller simply hears a fast busy (reorder) If I do a reload at the command prompt all is well for another few days..... What I'm