search for: flowroute

...o 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 back to SIP and the problem still occurred, so that was not it. We connect to Flowroute for our SIP provider.  I added "force_avp = yes" to the Flowroute endpoint section in the pjsip.conf and the problem appeared to be solved after I tested it a dozen times.  However, this morning this issue has reappeared.  Any thoughts on what might be causing this? My Flowroute pjsip.co...

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 supp...

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

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 seconds, some callers think they are being...

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: <http://lists.digium.com/pipermail/asterisk-users/attachments/20121129/83c846a4/attachment.htm>

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 not consistent though. I was on the phone with an insurance company yesterday for about 1 hour and the call was perfect...

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 also am not sure what or how I can connect a Network Only...

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 MessageSend <http://www.anveo.com/faq.asp?cod...

...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 Flowroute. The setting are the same as Vitelity. And amazingly, this DID works perfectly. This to me would indicate the problem whatever it is, is Vitelity or its upstream provider. - Am I right here?? Also, I have other DID blocks that have been with Vitelity for a while that do not have this issues....

...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 forwarded via SIP to my asterisk 1.6.2.4 server. The dtmfmode is set to rfc28333 in sip.conf. The first time this happened, I figured the caller pressed the number twice without realizing it. It's happening to too many people for that to be plausible anymore. I also experienced i...

Hello, I'm having a problem trying to do this, and it used to work with Asterisk 1.4. Since Asterisk 1.6 series I have not been able to place more than one call to a DID. I get this message: Skipping dialing interface 'SIP/16034817531 at flowroute' again since it has already been dialed I'm trying to do this because many of my clients have roll-over lines and they need to receive more than one call at a time. Any ideas? Thanks, -Mike Frager -------------- next part -------------- An HTML attachment was scrubbed... URL: http:/...

...wever, we don't want to maintain the DSL line or deal with the hassles of analog/digital conversion any more. So we want to switch to a reliable VoIP provider and move the asterisk server to one of our colocation data centers. We've tried getting test accounts with three VoIP providers: FlowRoute, CallCentric, and Vitelity. In our tests, outbound calls now go from softphones -> asterisk -> VoIP provider -> outside world. We use ulaw all the way through. But with all three providers, we see a curious thing: The audio quality in the direction from our softphones to the outside w...

> 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 MessageSend > <http://www.an...

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

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

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 that the call file creates the first part of the call, then creates a second call with the Dial() app. Once the call executed by the Dial() app is answered, the two calls are joined together. What I...

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 recipi...

We have a set (about 20) of DID's that we're not using. No one calls them, and we don't need them for outgoing. I'd like to keep them for future use. We now pay $5/mo/DID to host them. Is there a way to "warehouse" them? Just put them in a bank someplace? Thanks, sean

...xxxx at newyork.voip.ms' timed out, trying again (Attempt #1) [Jun 21 01:51:46] NOTICE[13657]: chan_sip.c:22943 sip_poke_noanswer: Peer '1850' is now UNREACHABLE! Last qualify: 15 [Jun 21 01:51:46] NOTICE[13657]: chan_sip.c:11569 sip_reg_timeout: -- Registration for 'xxxxxxx at sip.flowroute.com' timed out, trying again (Attempt #1) [Jun 21 01:52:07] NOTICE[13657]: chan_sip.c:22943 sip_poke_noanswer: Peer '1800' is now UNREACHABLE! Last qualify: 7 [Jun 21 01:52:07] NOTICE[13657]: chan_sip.c:22943 sip_poke_noanswer: Peer '1801' is now UNREACHABLE! Last qualify: 11...