similar to: SIP providers USA

Hello I like to make calls to an h.323 device. I'm using Nuphone h323. Compiled everything okay "I Guess" When I make a connection * SIP > h323 device, the phone is ringing and then * tells me "No one available....." and disconnect Thinking this is a codec problem and check in CLI h.323 show codecs and * shows nothing. I try many combination in the h323.conf like.

Hello Does anybody who have experience in installing the h323 modules in asterisk. I try'd it many times an spend xxx hours to install it but didn't get lucky so far. I have asterisk 1.0.1 running with bristuffed 2.0 I'm willing to pay for this. Sjaak

Hello * Users Did somebody get managed to get AreskiCC work under mysql. If so is there anywhere to find the database structure for mysql. Thanks Sjaak

Hello I like to call to 2 providers provider X = IAX provider Y = SIP exten => _06.,1,Dial(IAX2/X/${EXTEN},30,r)&(SIP/${EXTEN}@Y.Y.Y.Y) exten => _06.,2,Hangup Provider X is working but provider Y never shows up. What's wrong ?? How can I get provider Y working a fraction earlier the provider X Thanks Sjaak

Hello everyone I'm using rsync over ssh rsync -ave ssh bla@bla.tld:/home /backup/server1 This works great for many years now Now I'm playing with Nagios and NSCA but how can I detect if rsync has done everything well. nsca works simple <hostname>[tab]<svc_description>[tab]<return_code>[tab]<plugin_output>[newline]. I have a text file named backup_okay with

Hello * users I've problems with sound quality on zaphfc Asterisk works fine good sound quality. If I do "make load" in the bristuf.xx zaphfc dir then sound quality drops directly. Even if I don't load the chan_zap in the modules.conf I use this config on more (even old 400Mhz machines) and works correctly. Looks like an hardware problem but I can't find it. I don't

Hi list! I am installing an * box that will be installed on a site with KPN BRI ISDN in The Netherlands. I am using bristuff fron Junghanns. Does anybody know the correct settings for this? I will not have internet access there which makes it harder to google around on location. switchtype = euroisdn is pretty obvious but what about these settings: signalling = bri_cpe_ptmp ; p2p TE mode

Hi all, I've been playing about with Asterisk for years now on and off, just SIP to SIP calls, using FWD and suchlike. I'm moving house at the beginning of September and have decided to build an Asterisk based system for my home office. I'm in the UK and wonder if anyone can give me advice on lines and hardware to use. Had planned to go with an ISDN2e line coupled with a BT

Hi all i was going through the documents i need to achive the following setup, but iam confused to deploy but some one recomed me what will be good ------------------------ offic ------------------------ other office ----- Switch ----lan users ___________________ --- fiber link ___ wireless link now i want fiber both the links to be load balance from other office to this

I am going to try and use a passive AVM fritz BRI card for my * setup.. Here is the thing.. I need to order my BRI from BT.. The service that looks to be the one to use is what they call ISDN 2e becasue this has the option to setup hunt groups across multiple ISDN2e lines so I could add another line later to get 4 channels.. According to the BT website in order to use the hunt grouping across

Hey, I have free calling to between DDIs and cellphones on our group plan. I figure it'd be nice to allow staff with those cellphones to be able to forward callers to their VoiceMail to their cellphones using the * feature. I have a standard extension macro that has VoiceMail support. So far I've done this by duplicating the standard extension macro, and adding this rule (where ARG1 is

Hi all i have just set up a asterisk in china, using DE410P and one E1 line and get a phone number like: +86 020 87654321 from my sp when somebody dial +86 020 87654321 , the asterisk will get the call in number by ${EXTEN} variable, but it can only get 87654321, no area code . when someone dial +86 020 87654321xxxx , xxxx means 4 digits, the phone can call in, and the

Hi all, I have an asterisk-server (cvs-head from august) connected to a carrier's switch (DMS/Euroisdn) via a te410p, and I am having problems with DDI (standard 'official pstn' number plus extra digits for 'internal' use) Basically, when the entire number (including the extra digits) is dialled via a redial or a programmed key, I see the entire called party number (including

Hello I buyed a new server 2*XEON in a 2inch High 19"case. Now I have a problem that the riser card is 64bit so an ISDN PCI modem isn't possible. My question is can I use ISDN on com port /dev/ttyS0. If yes can I use it like the example in modem.conf as /dev/ttyI0 but use /dev/ttyS0 Does anybody have expirience with this ? Thanks Sjaak -- Dit bericht is gescand op virussen en

Hello Would it be possible to dail out to lett's say to 4 people with a .call file and put them directly into a free meetme room. Thanks Sjaak

Hi, When a VOIP user dials an external number, the calls are routed through our SIP provider. Is there a simple way to check whether the DDI exists locally before dialling out to the sip provider? Something like GotoIfExists(5551234 at incoming_calls) Currently, I'm paying for all calls, regardless of whether they exist locally. All DDIs exist in the incoming_calls context. Thanks Dan

Interested to know a good wholesale sip providers for 15k concurrent calls regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20231120/75b0e652/attachment.html>

On Monday 20 November 2023 at 12:14:11, Tahir Almas Dhesi wrote: > Interested to know good wholesale SIP providers for 15k concurrent calls You might want to specify a bit more detail, such as: - which country are you located in - do you require inbound DDIs (if so, in which region/s)? - which countries' Caller ID/s do you need to present? Antony. -- These clients are often

Interesting problem: An organization has departments. Each department has a single phone. Each department has multiple people. Each person within the organization has a direct dial incoming number. It's easy to set * up so that multiple DDIs get mapped to the same extension. What I'm wondering is if there's any way, with reasonably priced hardware, to notify the person who's

Hi Friends, I want to setup multiple SIP accounts. How can I do this? I have installed Asterisk, created Asterisk SIP extensions and registered in www.sipgate.co.uk. Now, what I have to do? 1) Am I need to install SER or OpenSER in my server along with Asterisk? 2) If yes, can you please recommond SER or OpenSER? 3) I searched in Internet. But, I didn't find good tutorial for this. Can you