similar to: Unable to place 2 or more calls to a DID

I upgraded our Asterisk 13 LTS to Asterisk 16 Certified 16.8 yesterday and converted form SIP to PJSIP using the python script as a start and then mofiying from there.  I ran into an issue when testing that incoming calls from MagicJack would go silent after about 10 seconds.  This happened while in the automated attendant area.  This problem did not occur with Asterisk 13 LTS.  I reverted PJSIP

Does anybody have T.38 faxing working with Flowroute? I am running Asterisk 1.6.2.7-rc3 with spandsp 0.0.6-pre17. I can successfully receive faxes over ulaw. I enabled T.38 with t38pt_udptl=yes in sip.conf. When I receive a fax it tries to negotiate T.38 and Flowroute sends back a Bad Request response saying I have a SIP syntax error. Flowroute support is recommending that I try again after

Asterisk 1.4.32 dahdi-2.3.0.1 Centos 5.5 Digium TE420 CAC channel bank (2) Cisco RVS4000 router Cox 50 Mbps/ 5 Mbps cable modem Flowroute provider codac G-711 90 % CPU idle callwaiting=no When there are 10-15 or more calls up the farend hears a callwaiting like beep every 3 to 6 sec. the duration of this "beep" is very short and would be no problem if it didn?t happen every few

I have an Asterisk 1.4.2 system that has been installed for about 3 months now in our home. We converted all of our phones to SIP phones, and use two different trunk providers (BroadVoice for incoming & FlowRoute for outgoing). Most of the time its working flawlessly. But about 1/3rd of the calls that come into us complain of an echo and what is best described as latency issues. Its

Hello List, Since I'm looking for a new VoIP provider for US origination/termination, I will very appreciate if you can chare your experience with Flowroute, Vitelity and Voip.ms Thanks in advance! Elder D. Arohuanca dCAP 1497 Lima - Peru -------------- next part -------------- An HTML attachment was scrubbed... URL:

What is the best way for me to setup Fax Capability with VOIP only. I have a Asterisk Server hosted on the internet without a modem. I'm using Flowroute, which is working awesome, for VOIP calls. I only have a SIP Phone at home and two Printer/Scanner/Fax Printers. I'm not sure which Fax Addons or Extensions I should use for Asterisk. I'd like it to Self Detect on any line. I

Can anyone comment on using SMS in conjunction with VoIP service using one of these three VoIP providers: voip.ms, vitelity.com, flowroute.com? Are some SMS services more compatible with Asterisk (i.e. SMS over SIP works perfectly or not)? Is it best to use a different data channel for SMS messages (i.e. SMS via HTTP, SMS via XMPP) instead of Asterisk's built in SMS application

Hi, I've been getting complaints lately that callers to my IVR are pressing a digit once but the system is responding as if they pressed it twice (once for each of two consecutive menus). I'm using an AGI script and logging all DTMF entries - and to the script, at least, it looks like the digit is being pressed twice. The TN being called is a VOIP number (provided by Flowroute) and being

I have a block of DID's that I ported to Vitelity about 7 days ago. The problem is if a POTS caller dials into the system, his dtmf is not heard at READ() or Background() while a prompt is played. After the prompt is finished, then dtmf is heard. I've been working with their support, but it still not resolved. SIP callers are not effected. Yesterday, I purchased a DID from

All, I'm trying to send an SMS directly from asterisk but it doesn't seem to be working. The SMS() function does create an outgoing file but doesn't deliver the SMS. Can anyone help me to understand how SMS() works. Thanks. extensions.conf example: same => n,SMS(hello,a,17654307001,"hello nick") - nick

We currently use asterisk 1.4.x with two Zaptel cards connected to POTS lines. So we make "outbound" calls from their softphones (using ulaw format), which go over a dedicated DSL line to the asterisk server in our office, which then converts the calls to POTS. This all works fine, assuming there aren't any unusual problems. It sounds as good as POTS on both ends. However, we

> Can anyone comment on using SMS in conjunction with VoIP service using > one of these three VoIP providers: voip.ms, vitelity.com, > flowroute.com? Are some SMS services more compatible with Asterisk > (i.e. SMS over SIP works perfectly or not)? Is it best to use a > different data channel for SMS messages (i.e. SMS via HTTP, SMS via > XMPP) instead of Asterisk's built

Hi, Anyone knows a DID provider that can do both outbound and inbound? Regards Nhadie

Hi, I am using a call file formated like this: Channel: local/12125557891 at outbound/n Callerid: 12125551212 Context: detect Extension: s Priority: 1 This sends the call into the dialplan at the [outbound] context. In [outbound], I have: [outbound] exten => _1.,1,Dial(SIP/${EXTEN}@flowroute,43) If the call is answered, it move on to the [detect] context. When using this method, it appears

Thanks to "WipeOut" who told me how to install the AVM fritz card. I think asterisk is working, no errors or warning by starting asterisk. Now I'm in trouble with the configuration files. I want to place calls over the ISDN network using an IP phone from Micronet. How can I setup the system? Thanks everybody who can help me.

The SIP trace will be adequate but this is on a remote system with limited disk space. I would love to turn on debugging while making the troublesome calls, then turn it off afterward. Tcpdump is great, but starting it and stopping it and keeping all that data would still be an issue. d On Fri, Feb 17, 2017 at 4:56 PM, Tim Pozar <pozar at lns.com> wrote: > Why not capture the packets

Hi I need help This is the error: Really destroying SIP dialog 'NDMxOWRmYTRhMWVkMGFhMjllMzU4YmNmNjQwN2NlM2Y.' Method: SUBSCRIBE -- Executing [00919885497796 at internal:1] Set("SIP/1001-0000000b", "CALLERID(num)=8790771141") in new stack -- Executing [00919885497796 at internal:2] Dial("SIP/1001-0000000b", "SIP/00919885497796 at sonetel")

On 8/27/10, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body

We recently completed a full migration of our office phone system over to Asterisk and are using flowroute & Broadvoice as SIP providers for our incoming/outgoing calls. Everything is working great. Now that the phones are done, I'm left with the fax machine. Although we have been using jFax for digital fax handling for most of our fax needs, it falls short with many calls where the

I'm wondering if any other Asterisk users have a recommendation for a reliable SIP Trunk provider that supports Asterisk and offers decent support. I've worked with Coredial, Broadvox, and Broadvoice and have had some bad experiences with each of these providers. Broadvoice offers low cost service, however I have constant issues with Broadvoice blocking my customers due to Asterisk