# Problem with calling Out **URL:** \ **Category:** Asterisk Support\ **Created:** [March 23, 2007, 11:38am UTC](https://community.asterisk.org/t/problem-with-calling-out/13296 "2007-03-23T11:38:13Z")\ **Posts on this page:** 16\ **Page:** 1
**Author:** ![arosa](https://avatars.discourse-cdn.com/v4/letter/a/e95f7d/32.png) [@arosa](https://community.asterisk.org/u/arosa)\ **Post date:** [March 23, 2007, 11:38am UTC](https://community.asterisk.org/t/problem-with-calling-out/13296/1 "2007-03-23T11:38:13Z")
Hi, I tried to configure my sip provider with the web interface, I check the user.conf, and tried everyting and nothig work. I check if my provider register me and the sip show registry command told me that I’m registered. Here is the error I got when I tried to place an outside call: – Executing [(THE NUMBER I DIAL)@numberplan-custom-1:1] Macro(“SIP/227-08210278”, "trunkdial|SIP/trunk\_1/( The number I dial )in new stack – Executing [s@macro-trunkdial:1] Dial(“SIP/227-08210278”, “SIP/trunk\_1/7874350473”) in new stack – Called trunk\_1/7874350473 [Mar 23 07:26:29] WARNING[5367]: chan\_sip.c:11731 handle\_response\_invite: Received response: “Forbidden” from ‘“Comp” \;tag=as5c282715’ – SIP/trunk\_1-08213288 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) – Executing [s@macro-trunkdial:2] Goto(“SIP/227-08210278”, “s-CONGESTION|1”) in new stack – Goto (macro-trunkdial,s-CONGESTION,1) – Executing [s-CONGESTION@macro-trunkdial:1] NoOp(“SIP/227-08210278”, “”) in new stack My trunk in the user.conf is this [trunk\_1] secret = (MY PASSWORD) provider = trunkstyle = customvoip username = (MY USER NAME) trunkname = Custom - MY PROVIDER NAME callerid = (MY SIP PHONE NUMBER) hasexten = no hassip = yes hasiax = no registeriax = no registersip = yes host = (MY SIP PROVIDER) dialformat = ${EXTEN:1} context = DID\_trunk\_1 group = insecure = fromuser = (MY SIP PHONE NUMBER) fromdomain = With my other provider(Stanaphone) work perfectly, I don’t known why this new one don’t work. Any idea? ---
**Author:** ![arosa](https://avatars.discourse-cdn.com/v4/letter/a/e95f7d/32.png) [@arosa](https://community.asterisk.org/u/arosa)\ **Post date:** [March 24, 2007, 12:42pm UTC](https://community.asterisk.org/t/problem-with-calling-out/13296/2 "2007-03-24T12:42:33Z")
Hi to all, now I can place an outside call, my regular phone ring and I answer the phone but I can’t hear nothing after few seconds the call is dropped. I don’t know if my problem is the type of codec i use. This is part of my sip configuration for my sip provider; [SIP PROVIDER] type = peer ;user = phone host = MY SIP PROVIDER fromdomain = MY SIP PROVIDER fromuser = MY SIP PHONE secret= MY PASSWORD username= MY USERNAME insecure = very context = MY PROVIDER authname = MY USERNANE dtmfmode = inband dtmf = inband ;Disable canreinvite if you are behind a NAT canreinvite = no Any Idea ? ---
**Author:** ![parkley](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/parkley/32/839_2.png) [@parkley](https://community.asterisk.org/u/parkley)\ **Post date:** [March 27, 2007, 10:21pm UTC](https://community.asterisk.org/t/problem-with-calling-out/13296/3 "2007-03-27T22:21:43Z")
Try playing around with the NAT setting. Makes a lot of difference for some phone models. Rgds Patrick Arkley [www.say-no.se](http://www.say-no.se) ---
**Author:** ![arosa](https://avatars.discourse-cdn.com/v4/letter/a/e95f7d/32.png) [@arosa](https://community.asterisk.org/u/arosa)\ **Post date:** [March 27, 2007, 11:11pm UTC](https://community.asterisk.org/t/problem-with-calling-out/13296/4 "2007-03-27T23:11:46Z")
Thanks for the answer, the problem was the Nat, now is working ---
**Author:** ![rasolrac](https://avatars.discourse-cdn.com/v4/letter/r/5f9b8f/32.png) [@rasolrac](https://community.asterisk.org/u/rasolrac)\ **Post date:** [March 28, 2007, 1:17am UTC](https://community.asterisk.org/t/problem-with-calling-out/13296/5 "2007-03-28T01:17:56Z")
Hi Arosa Please could you explain how do you resolved the problem for the outgoing calls? I have the same problem and i can’t resolve it thanks ---
**Author:** ![tknman0700](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/tknman0700/32/1098_2.png) [@tknman0700](https://community.asterisk.org/u/tknman0700)\ **Post date:** [March 28, 2007, 2:07pm UTC](https://community.asterisk.org/t/problem-with-calling-out/13296/6 "2007-03-28T14:07:28Z")
I am also having this same problem… any post you could give would be appreciated. ---
**Author:** ![baconbuttie](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/baconbuttie/32/916_2.png) [@baconbuttie](https://community.asterisk.org/u/baconbuttie)\ **Post date:** [March 28, 2007, 2:09pm UTC](https://community.asterisk.org/t/problem-with-calling-out/13296/7 "2007-03-28T14:09:50Z")
NAT issues are covered pretty extensively on the wiki at [voip-info.org](http://www.voip-info.org) and on other pages (search google for “asterisk nat”) or if you search this forum you’ll find the same question asked and answered many many times. ---
**Author:** ![tknman0700](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/tknman0700/32/1098_2.png) [@tknman0700](https://community.asterisk.org/u/tknman0700)\ **Post date:** [March 28, 2007, 2:11pm UTC](https://community.asterisk.org/t/problem-with-calling-out/13296/8 "2007-03-28T14:11:46Z")
Are you indicating a nat issue between the phone and the A box or are you saying between the A box and the sip provider? ---
**Author:** ![baconbuttie](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/baconbuttie/32/916_2.png) [@baconbuttie](https://community.asterisk.org/u/baconbuttie)\ **Post date:** [March 28, 2007, 2:29pm UTC](https://community.asterisk.org/t/problem-with-calling-out/13296/9 "2007-03-28T14:29:36Z")
it doesn’t really matter … if there’s a NAT between Asterisk and a peer or user then this needs to be accounted for in the configs. ---
**Author:** ![parkley](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/parkley/32/839_2.png) [@parkley](https://community.asterisk.org/u/parkley)\ **Post date:** [March 28, 2007, 4:44pm UTC](https://community.asterisk.org/t/problem-with-calling-out/13296/10 "2007-03-28T16:44:31Z")
Well, it does matter. I had 2 Polycom 501 and 4 Cisco 7940 on a private network. The Asterisk where also on the same lan. Regardless of having NAT set to yes or no for the Polycom peers the Polycoms could register. The Ciscos couldn’t. When I set the Cisco peers to NAT=no they could register as well. I’m not sure thid is covered on the NAT section. Rgds Patrick Arkley [www.say-no.se](http://www.say-no.se) ---
**Author:** ![baconbuttie](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/baconbuttie/32/916_2.png) [@baconbuttie](https://community.asterisk.org/u/baconbuttie)\ **Post date:** [March 28, 2007, 4:48pm UTC](https://community.asterisk.org/t/problem-with-calling-out/13296/11 "2007-03-28T16:48:41Z")
> [@baconbuttie](#): > > if there’s a NAT between Asterisk and a peer or user then this needs to be accounted for in the configs. ---
**Author:** ![parkley](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/parkley/32/839_2.png) [@parkley](https://community.asterisk.org/u/parkley)\ **Post date:** [March 28, 2007, 7:23pm UTC](https://community.asterisk.org/t/problem-with-calling-out/13296/12 "2007-03-28T19:23:47Z")
> [@baconbuttie](#): > > [quote=“baconbuttie”]if there’s a NAT between Asterisk and a peer or user then this needs to be accounted for in the configs.[/quote] Sorry, was a bit trigger happy 😉 /Patrick Arkley (ww.say-no.se) ---
**Author:** ![rasolrac](https://avatars.discourse-cdn.com/v4/letter/r/5f9b8f/32.png) [@rasolrac](https://community.asterisk.org/u/rasolrac)\ **Post date:** [March 28, 2007, 11:03pm UTC](https://community.asterisk.org/t/problem-with-calling-out/13296/13 "2007-03-28T23:03:55Z")
guys i’m not talking about the NAT problem, i’m talking about the Arosa’s first problem, so the post. (CONGESTION) 😃 ---
**Author:** ![baconbuttie](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/baconbuttie/32/916_2.png) [@baconbuttie](https://community.asterisk.org/u/baconbuttie)\ **Post date:** [March 29, 2007, 3:02am UTC](https://community.asterisk.org/t/problem-with-calling-out/13296/14 "2007-03-29T03:02:33Z")
> [@arosa](#): > > Thanks for the answer, the problem was the Nat, now is working this is what the OP actually said ---
**Author:** ![arosa](https://avatars.discourse-cdn.com/v4/letter/a/e95f7d/32.png) [@arosa](https://community.asterisk.org/u/arosa)\ **Post date:** [March 30, 2007, 10:11am UTC](https://community.asterisk.org/t/problem-with-calling-out/13296/15 "2007-03-30T10:11:30Z")
Hi to all in the forum, to resolve the problem I got, the first thing I did was to place the Asterisk Server in the DMZ zone in the router to eliminate some NAT problem, then I create my sip provider with the web interface. Then I modify the user.conf, where the sip provider is created. The interesting part is when I create my sip provider with the web interface I need to fill some settings, like the (provider = My Provider , insecure = very, fromdomain=My Provider). Then when I test and everything is working I remove the server from the DMZ and configure the ports the asterisk server needs, (UDP Ports 5060, 10000-20000) [trunk\_1] secret = (MY PASSWORD) provider = trunkstyle = customvoip username = (MY USER NAME) trunkname = Custom - MY PROVIDER NAME callerid = (MY SIP PHONE NUMBER) hasexten = no hassip = yes hasiax = no registeriax = no registersip = yes host = (MY SIP PROVIDER) dialformat = ${EXTEN:1} context = DID\_trunk\_1 group = insecure = fromuser = (MY SIP PHONE NUMBER) fromdomain = ---
**Author:** ![rasolrac](https://avatars.discourse-cdn.com/v4/letter/r/5f9b8f/32.png) [@rasolrac](https://community.asterisk.org/u/rasolrac)\ **Post date:** [April 4, 2007, 12:28am UTC](https://community.asterisk.org/t/problem-with-calling-out/13296/16 "2007-04-04T00:28:45Z")
Thanks Guys but i still have the problem… 👿👿☹☹ I don’t know what else to try.