Hi, perhaps someone can point me int he right direction. My SIP are registered on all trunks, including the MagicJack trunks. However, I cannot call out on the MagicJack trunks anymore since I upgraded to 2.70 today. The other trunks work fine.
I get : "the number you have dialed is not in service… " from Asterisk then the busy signal
Thanks for your help.
Thank you for pointing that out. I must have missed that when setting up the system. As a Newbie to both Linux and FreePBX, I ahve to say that there was a lot of reading to do. Coming from MS environment (I go back to DOS 3.0) the lerning curve is steep in the beginning. I might suggest that there be a Linux for newbies section in the community because even with a DOS background from 20+ years agao, the commands are not the same and it takes some time to figure things out. IF there is a Linux for Newbie section, somehow I could not find that.
My hats are aoof to you Phillippe and all the rest (Mikael especially too)for all the tireless work you have done. When I deploy FreePBX in the work environment, I will decide to use SIP Station instead of all the work-arounds I have done for my home and personal environment. Please keep up the great work and I hope I can contribute somehow to the community more.
You may want to provide a bit more information on your issue, such as a CLI trace with verbosity set to 5 when making the failing calling.
I just upgraded to 2.6 and lost all my outbound calls.
In actual fact, I lost the content of all my Outbound Routes.
Please jono, what you posted is called a “hijack of a thread”. Your question has nothing at all to do with the original question.
Please create a new thread with your question and post more relevant information such as what distro you use, from what version of FreePBX you upgraded from etc etc.
Do NOT reply to this thread!
Watcha on about Willis?
Please read the OP, then mine. I didn’t even ask a question.
Can you tell me how to do a CLI trace and set the vebosity to 5? Sorry to ask, I am new to Asterisk
Had you previously patched your chan_sip to work with Magicjack before the upgrade? If so you probably need to patch the upgraded version as well.
No I didn’t patch it, I have been using the work-around by utilizing the MJMD5 which I believe handles the port forwarding.
This is an internal error in Asterisk I believe since I get the Asterisk “the number you ahve dialed is not in service …” and it does not apear to connect through a SIP channel, but I cannot be sure. If someone could point em int he direction of what I need to do to execute a trace of some sort, iw ill be happy to provide it.
On what machine are you running the MJMDM5 proxy? Have you changed any host settings in your trunk? I use the patch but I would think you’d need your host to be the ip of your MJMDM5 proxy.
I tracked down the issue and it was SIP settings necessary for Viatalk to work on outgoing.I had the ViaTalk SIP settings in the additional fields under Asterisk SIP Settings. Though I want to make it clear that before the upgrade the two SIP trunks co-habitated nicely together with the SIP settings for ViaTalk.
The solution was to take out most of the SIP settings neccessary for ViaTalk to work from the Asterisk SIp Settings, and place those in the ViaTalk Trunk configuration for the outbound.Once I did this, both trunks works properly in and out. Perhaps this is a lesson that should be posted somewhere that if a SIP provider requires certain settings, that it should be listed in the Trunk Outbound Configuration…??.
I would like to offer for anyone that is wanting to make MagicJack work on their Asterisk to go ahead and email me - I can tell you that early on it was frustrating to get MagicJack to work properly, but now it works so well, I use those as the primary outbound trunks. I have the least amount of problems with those trunks, they connect immediately, and never have jitter. I will say though for Fax they are not reliable and I have routed my fax on the ViaTalk which does a great job.
All in all, FreePBX is an amazing product, and I am amazed at the amount of add-on’s, and good software that went into this product. Though I am not a Linux person (at least before this), I am now a fan of the operating system.
provider specific SIP settings always belong with the trunk. The Asterisk SIP Settings module is nothing more than a ‘handy’ way to enter settings into the SIP [general] section. Furthermore, there was not change at all to the way SIP settings are generated between 2.6 and 2.7, in trunks or that module.
Upgrades to Asterisk can effect changes sometimes if by chance you upgraded at the same time. Otherwise this appears to have been some other change that occurred at the same time as your upgrade because the sip*.conf file generation is all identical.
Glad you got your issue resolved though.
This wasn’t a Linux issue nor a FreePBX issue.
People somehow perceive that FreePBX relieves you of the responsibility of understanding how Asterisk works.
The concept of peer vs. genral settings is an Asterisk concept. It is the same throughout all channel technologies including SIP.
Asterisk, The Future of Technology www.asteriskdocs.org (buy a hard copy from Amazon) is a must read for anyone serious about deploying an Asterisk based solution.
FreePBX simplifies many administrative tasks. Trunks (which are really peers) allow you to use any Asterisk variable. Voip-info.org has great pages on all the declaratives and syntax. They are a great reference.
As far as the sysadmin stuff, you can Google almost any administrative task and somewhere on the web someone has document how to do it. So many good Linux books exist I don’t know where to start.