Re: [asterisk-users] Outbound Calls via Proxy to use Call ID from registration

[Joshua Colp]

On Mon, Aug 28, 2017, at 05:45 AM, Benoit Panizzon wrote:
> Hello List
> 
> > I work at an SIP Provider and we have added and SBC in front of our
> > Voice Switch to protect it.
> 
> Well using two peers for incomming and outgoing calls solve the
> previous issue.
> 
> Now I have a new one.
> 
> The SBC in use needs to match incomming calls from the asterisk with
> the call id used in the registration.
> 
> We have tested this with a couple of PBX, which do use the call ID used
> during registration automatically for outbound invites.
> 
> Not so my asterisk server.
> 
> So I assumed that when I refer to a 'peer' definition in the register
> statement, I could make asterisk understand, that the registration and
> outgoing peers belong together and then use the same call ID.

Can you define what exactly you mean by call id? If you are referring to
the Call-ID SIP header that's not how it works. It's unique within a
dialog and not reused like that[1][2].

[1] https://tools.ietf.org/html/rfc3261#page-37
[2] https://tools.ietf.org/html/rfc3261#section-20.8

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] outbound calls

[Salaheddine Elharit]

hi



the issue still the same i have 2 trunks whe i configure the first in
x-lite and the second in my server or my ip-phone snom320 directly



from x-lite i can call my trunk without issue but when i try ti call from
snom320 to x-lite or from my server asterisk using extension in x-lite the
call all time is failed



any help please



thanks and regards

2015-03-20 19:28 GMT+00:00 Trey Hilyard :

> So you are saying that it resolved the issue to activate voicemail on the
> device that sits past your trunk provider? That confuses me a little, but
> if your calls are working, that's great news.
>
> On Fri, Mar 20, 2015 at 1:44 PM Salaheddine Elharit <
> salah.elharit...@gmail.com> wrote:
>
>> i noticed that when i active the voicemail in the IP-phone where the
>> number 0033149xx is configured i can call this number without issue
>>
>> Using SIP RTP TOS bits 184
>>   == Using SIP RTP CoS mark 5
>> -- Called SIP/FD/0033149xx == Begin MixMonitor Recording
>> SIP/101-010d
>> -- SIP/FD-010e is making progress passing it to SIP/101-010d
>>> 0x2b393cfc2610 -- Probation passed - setting RTP source address
>> to 192.
>>168.1.138:55542
>>> 0x1d08efa0 -- Probation passed - setting RTP source address to
>>  217.195.xx.xx:46346
>> -- SIP/FD-010e answered SIP/101-010d
>>> 0x1d08efa0 -- Probation passed - setting RTP source address to
>>  217.195.xx.xx:46346
>> thanks and regards.
>>
>>
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Re: [asterisk-users] outbound calls

[Salaheddine Elharit]

thanks for your response

i noticed that when i active the voicemail in the IP-phone where the number
0033149xx is configured i can call this number without issue

the server asterisk and the ip-phone where the number is configured are in
the same network 192.168.1.X

Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
-- Called SIP/FD/0033149xx
  == Begin MixMonitor Recording SIP/101-010d
-- SIP/FD-010e is making progress passing it to SIP/101-010d
   > 0x2b393cfc2610 -- Probation passed - setting RTP source address to
192.
 168.1.138:55542
   > 0x1d08efa0 -- Probation passed - setting RTP source address to
 217.195.xx.xx:46346
-- SIP/FD-010e answered SIP/101-010d
   > 0x1d08efa0 -- Probation passed - setting RTP source address to
 217.195.xx.xx:46346
thanks and regards.

2015-03-20 18:39 GMT+00:00 Salaheddine Elharit :

> thank you
>
> i noticed that when i active the voicemail in the IP-phone where the
> number 0033149xx is configured i can call this number without issue
>
> Using SIP RTP TOS bits 184
>   == Using SIP RTP CoS mark 5
> -- Called SIP/FD/0033149xx == Begin MixMonitor Recording
> SIP/101-010d
> -- SIP/FD-010e is making progress passing it to SIP/101-010d
>> 0x2b393cfc2610 -- Probation passed - setting RTP source address
> to 192.
>168.1.138:55542
>> 0x1d08efa0 -- Probation passed - setting RTP source address to
>  217.195.xx.xx:46346
> -- SIP/FD-010e answered SIP/101-010d
>> 0x1d08efa0 -- Probation passed - setting RTP source address to
>  217.195.xx.xx:46346
> thanks and regards.
>
> 2015-03-20 17:15 GMT+00:00 Trey Hilyard :
>
>> I am making some assumptions, but assuming the 217.195.xx.xxx is your
>> provider, you are getting this back from them:
>>
>> "Got SIP response 556 "No address found" back from 217.195.xx.xxx:5060"
>>
>> Are you sure that "0033149xx" is the format the provider is
>> expecting? You might try enabling SIP debug on the 217.195.xx.xx IP and
>> seeing what the INVITE looks like, but normally a 556 indicates that your
>> provider didn't have routing for either the R-URI or they didn't recognize
>> that is was coming from you. You might compare the SIP INVITE coming from
>> Asterisk to the one from Z-Lite and see where the differences are.
>>
>>
>>
>> On Fri, Mar 20, 2015 at 12:03 PM Salaheddine Elharit <
>> salah.elharit...@gmail.com> wrote:
>>
>>> hello list
>>>
>>> i have an issue related to outbound calls i can contact all the number
>>> except on number given by our provider in trunk
>>>
>>> the issue just when i configure my trunk in our server but when i
>>> configure the trunk directly in x-lite i can contact this number without
>>> issue
>>>
>>> below the cli
>>>
>>>   == Using SIP RTP TOS bits 184
>>>   == Using SIP RTP CoS mark 5
>>> -- Executing [0149xx@from-internal:1] Macro("SIP/101-0103",
>>> "user-callerid,LIMIT,EXTERNAL,") in new stack
>>> -- Executing [s@macro-user-callerid:1] Set("SIP/101-0103",
>>> "TOUCH_MONITOR=1426869820.301") in new stack
>>> -- Executing [s@macro-user-callerid:2] Set("SIP/101-0103",
>>> "AMPUSER=101") in new stack
>>> -- Executing [s@macro-user-callerid:3] GotoIf("SIP/101-0103",
>>> "0?report") in new stack
>>> -- Executing [s@macro-user-callerid:4] ExecIf("SIP/101-0103",
>>> "1?Set(REALCALLERIDNUM=101)") in new stack
>>> -- Executing [s@macro-user-callerid:5] Set("SIP/101-0103",
>>> "AMPUSER=101") in new stack
>>> -- Executing [s@macro-user-callerid:6] GotoIf("SIP/101-0103",
>>> "0?limit") in new stack
>>> -- Executing [s@macro-user-callerid:7] Set("SIP/101-0103",
>>> "AMPUSERCIDNAME=101") in new stack
>>> -- Executing [s@macro-user-callerid:8] GotoIf("SIP/101-0103",
>>> "0?report") in new stack
>>> -- Executing [s@macro-user-callerid:9] Set("SIP/101-0103",
>>> "AMPUSERCID=101") in new stack
>>> -- Executing [s@macro-user-callerid:10] Set("SIP/101-0103",
>>> "__DIAL_OPTIONS=tr") in new stack
>>> -- Executing [s@macro-user-callerid:11] Set("SIP/101-0103",
>>> "CALLERID(all)="101" <101>") in new stack
>>> -- Executing [s@macro-user-callerid:12] GotoIf("SIP/101-0103",
>>> "0?limit") in new stack
>>> -- Executing [s@macro-user-callerid:13] ExecIf("SIP/101-0103",
>>> "1?Set(GROUP(concurrency_limit)=101)") in new stack
>>> -- Executing [s@macro-user-callerid:14] ExecIf("SIP/101-0103",
>>> "0?Set(CHANNEL(language)=)") in new stack
>>> -- Executing [s@macro-user-callerid:15] GotoIf("SIP/101-0103",
>>> "1?continue") in new stack
>>> -- Goto (macro-user-callerid,s,28)
>>> -- Executing [s@macro-user-callerid:28] Set("SIP/101-0103",
>>> "CALLERID(number)=101") in new stack
>>> -- Executing [s@macro-user-callerid:29] Set("SIP/101-0103",
>>> "CALLERID(name)=101") in new stack
>>> -- Executing [s@macro-user-callerid:30] Set("SIP/101

Re: [asterisk-users] outbound calls

[Trey Hilyard]

So you are saying that it resolved the issue to activate voicemail on the
device that sits past your trunk provider? That confuses me a little, but
if your calls are working, that's great news.

On Fri, Mar 20, 2015 at 1:44 PM Salaheddine Elharit <
salah.elharit...@gmail.com> wrote:

> i noticed that when i active the voicemail in the IP-phone where the
> number 0033149xx is configured i can call this number without issue
>
> Using SIP RTP TOS bits 184
>   == Using SIP RTP CoS mark 5
> -- Called SIP/FD/0033149xx == Begin MixMonitor Recording
> SIP/101-010d
> -- SIP/FD-010e is making progress passing it to SIP/101-010d
>> 0x2b393cfc2610 -- Probation passed - setting RTP source address
> to 192.
>168.1.138:55542
>> 0x1d08efa0 -- Probation passed - setting RTP source address to
>  217.195.xx.xx:46346
> -- SIP/FD-010e answered SIP/101-010d
>> 0x1d08efa0 -- Probation passed - setting RTP source address to
>  217.195.xx.xx:46346
> thanks and regards.
>
>
> --
> _
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Re: [asterisk-users] outbound calls

[Salaheddine Elharit]

i noticed that when i active the voicemail in the IP-phone where the number
0033149xx is configured i can call this number without issue

Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
-- Called SIP/FD/0033149xx == Begin MixMonitor Recording
SIP/101-010d
-- SIP/FD-010e is making progress passing it to SIP/101-010d
   > 0x2b393cfc2610 -- Probation passed - setting RTP source address to
192.
 168.1.138:55542
   > 0x1d08efa0 -- Probation passed - setting RTP source address to
 217.195.xx.xx:46346
-- SIP/FD-010e answered SIP/101-010d
   > 0x1d08efa0 -- Probation passed - setting RTP source address to
 217.195.xx.xx:46346
thanks and regards.
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Re: [asterisk-users] outbound calls

[Trey Hilyard]

I am making some assumptions, but assuming the 217.195.xx.xxx is your
provider, you are getting this back from them:

"Got SIP response 556 "No address found" back from 217.195.xx.xxx:5060"

Are you sure that "0033149xx" is the format the provider is expecting?
You might try enabling SIP debug on the 217.195.xx.xx IP and seeing what
the INVITE looks like, but normally a 556 indicates that your provider
didn't have routing for either the R-URI or they didn't recognize that is
was coming from you. You might compare the SIP INVITE coming from Asterisk
to the one from Z-Lite and see where the differences are.



On Fri, Mar 20, 2015 at 12:03 PM Salaheddine Elharit <
salah.elharit...@gmail.com> wrote:

> hello list
>
> i have an issue related to outbound calls i can contact all the number
> except on number given by our provider in trunk
>
> the issue just when i configure my trunk in our server but when i
> configure the trunk directly in x-lite i can contact this number without
> issue
>
> below the cli
>
>   == Using SIP RTP TOS bits 184
>   == Using SIP RTP CoS mark 5
> -- Executing [0149xx@from-internal:1] Macro("SIP/101-0103",
> "user-callerid,LIMIT,EXTERNAL,") in new stack
> -- Executing [s@macro-user-callerid:1] Set("SIP/101-0103",
> "TOUCH_MONITOR=1426869820.301") in new stack
> -- Executing [s@macro-user-callerid:2] Set("SIP/101-0103",
> "AMPUSER=101") in new stack
> -- Executing [s@macro-user-callerid:3] GotoIf("SIP/101-0103",
> "0?report") in new stack
> -- Executing [s@macro-user-callerid:4] ExecIf("SIP/101-0103",
> "1?Set(REALCALLERIDNUM=101)") in new stack
> -- Executing [s@macro-user-callerid:5] Set("SIP/101-0103",
> "AMPUSER=101") in new stack
> -- Executing [s@macro-user-callerid:6] GotoIf("SIP/101-0103",
> "0?limit") in new stack
> -- Executing [s@macro-user-callerid:7] Set("SIP/101-0103",
> "AMPUSERCIDNAME=101") in new stack
> -- Executing [s@macro-user-callerid:8] GotoIf("SIP/101-0103",
> "0?report") in new stack
> -- Executing [s@macro-user-callerid:9] Set("SIP/101-0103",
> "AMPUSERCID=101") in new stack
> -- Executing [s@macro-user-callerid:10] Set("SIP/101-0103",
> "__DIAL_OPTIONS=tr") in new stack
> -- Executing [s@macro-user-callerid:11] Set("SIP/101-0103",
> "CALLERID(all)="101" <101>") in new stack
> -- Executing [s@macro-user-callerid:12] GotoIf("SIP/101-0103",
> "0?limit") in new stack
> -- Executing [s@macro-user-callerid:13] ExecIf("SIP/101-0103",
> "1?Set(GROUP(concurrency_limit)=101)") in new stack
> -- Executing [s@macro-user-callerid:14] ExecIf("SIP/101-0103",
> "0?Set(CHANNEL(language)=)") in new stack
> -- Executing [s@macro-user-callerid:15] GotoIf("SIP/101-0103",
> "1?continue") in new stack
> -- Goto (macro-user-callerid,s,28)
> -- Executing [s@macro-user-callerid:28] Set("SIP/101-0103",
> "CALLERID(number)=101") in new stack
> -- Executing [s@macro-user-callerid:29] Set("SIP/101-0103",
> "CALLERID(name)=101") in new stack
> -- Executing [s@macro-user-callerid:30] Set("SIP/101-0103",
> "CDR(cnum)=101") in new stack
> -- Executing [s@macro-user-callerid:31] Set("SIP/101-0103",
> "CDR(cnam)=101") in new stack
> -- Executing [s@macro-user-callerid:32] Set("SIP/101-0103",
> "CHANNEL(language)=en") in new stack
> -- Executing [0149xx@from-internal:2] Set("SIP/101-0103",
> "MOHCLASS=default") in new stack
> -- Executing [0149xx@from-internal:3] Set("SIP/101-0103",
> "_NODEST=") in new stack
> -- Executing [0149xx@from-internal:4] Gosub("SIP/101-0103",
> "sub-record-check,s,1(out,0149xx,)") in new stack
> -- Executing [s@sub-record-check:1] Set("SIP/101-0103",
> "REC_POLICY_MODE_SAVE=") in new stack
> -- Executing [s@sub-record-check:2] GotoIf("SIP/101-0103",
> "1?check") in new stack
> -- Goto (sub-record-check,s,7)
> -- Executing [s@sub-record-check:7] Set("SIP/101-0103",
> "__MON_FMT=wav") in new stack
> -- Executing [s@sub-record-check:8] GotoIf("SIP/101-0103",
> "1?next") in new stack
> -- Goto (sub-record-check,s,11)
> -- Executing [s@sub-record-check:11] ExecIf("SIP/101-0103",
> "0?Return()") in new stack
> -- Executing [s@sub-record-check:12] ExecIf("SIP/101-0103",
> "0?Set(__REC_POLICY_MODE=)") in new stack
> -- Executing [s@sub-record-check:13] GotoIf("SIP/101-0103",
> "0?out,1") in new stack
> -- Executing [s@sub-record-check:14] Set("SIP/101-0103",
> "__REC_STATUS=INITIALIZED") in new stack
> -- Executing [s@sub-record-check:15] Set("SIP/101-0103",
> "NOW=1426869820") in new stack
> -- Executing [s@sub-record-check:16] Set("SIP/101-0103",
> "__DAY=20") in new stack
> -- Executing [s@sub-record-check:17] Set("SIP/101-0103",
> "__MONTH=03") in new stack
> -- Executing [s@sub-record-check:18] Set("SIP/101-0103",
> "__YEAR=2015") in new stack
> -- 

Re: [asterisk-users] outbound calls via google voice not answered by toll free numbers with ivrs

[Gaurav P]

Apologize for following up to my own question, but wanted to mention that
some toll free numbers with ivrs work fine. Only run into issues with
certain numbers like the test number in my previous email.

Any ideas?

On Fri, May 13, 2011 at 10:26 AM, Gaurav P <
gaurav.lists+asterisk-us...@gmail.com> wrote:

> Hi All,
>
> I'm using Asterisk 1.8.2 with outbound calls using Google Voice. I've been
> having issues calling several toll free numbers where the call 'is ringing'
> but never transitions to 'answered'. These are toll free numbers which are
> typically answered by an ivrs where you enter eg. a conference bridge
> number.
>
> I searched google and the closest reported issues I found are -
>
> https://issues.asterisk.org/view.php?id=18319 (on 1.6.x)
> and
> https://issues.asterisk.org/view.php?id=5266 (where the ibm support number
> listed does not work for my setup either)
>
> The number in the second ticket can be used as a test case - 800-426-7378- 
> and I'm hoping someone has run into this before.
>
> I have already tried both 'auto' and 'rfc2833' settings for dtmfmode and
> can provide any additional details about my setup.
>
> Thanks in advance!
> -Gaurav
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Re: [asterisk-users] outbound calls not ringing still

[Olle E. Johansson]

3 sep 2009 kl. 00.27 skrev John A. Sullivan III:

> On Wed, 2009-09-02 at 21:31 +, Ott Rose wrote:
>> i have posted this before but was unable to resolve it. i have some
>> new info so i figured i would try again. the trace from bandwidth.com
>> are below. they are telling me that the ip that is bold should be our
>> ip not bandwidth.com. i have changed every setting that i can see and
>> nothing fixes this. Where would i change this at? they cannot tell  
>> me.
>>
>> INVITE sip:+185993133...@216.82.224.202 SIP/2.0
>> Via: SIP/2.0/UDP 216.82.224.202:5060;branch=z9hG4bK3691b08c;rport
>> From:"8592192438";tag=as0707d433
>> To:
>> Contact:
>> Call-ID: 0f3bdcc9171ef53148e7bab413aea...@64.191.130.78
>> CSeq: 102 INVITE
>> User-Agent: Asterisk PBX
>> Max-Forwards: 70
>> Date: Wed, 02 Sep 2009 21:10:39 GMT
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>> Supported: replaces
>> Content-Type: application/sdp
>> Content-Length: 412
>>
>> v=0
>> o=root 3831 3831 IN IP4 216.82.224.202
>> s=session
>> c=IN IP4 216.82.224.202
>> t=0 0
>> m=audio 17050 RTP/AVP 0 8 3 101
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:3 GSM/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=silenceSupp:off - - - -
>> a=ptime:20
>> a=sendrecv
>> m=video 12426 RTP/AVP 31 34 103
>> a=rtpmap:31 H261/9
>> a=rtpmap:34 H263/9
>> a=rtpmap:103 h263-1998/9
>> a=sendrecv
>>
> 
> I know very little about how ringing works but are they providing any
> kind of status information to you? Do you need to furnish the ring if
> they are not? It seems to me I saw quite a few articles about  
> providing
> ring tone, what causes it to fail, and how to work around it.  I  
> assume
> you've searched for those already. Just a few thoughts - John

It's very hard to say much without your configurations at hand.

/O

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Re: [asterisk-users] outbound calls not ringing still

[John A. Sullivan III]

On Wed, 2009-09-02 at 21:31 +, Ott Rose wrote:
> i have posted this before but was unable to resolve it. i have some
> new info so i figured i would try again. the trace from bandwidth.com
> are below. they are telling me that the ip that is bold should be our
> ip not bandwidth.com. i have changed every setting that i can see and
> nothing fixes this. Where would i change this at? they cannot tell me.
> 
> INVITE sip:+185993133...@216.82.224.202 SIP/2.0
> Via: SIP/2.0/UDP 216.82.224.202:5060;branch=z9hG4bK3691b08c;rport
> From:"8592192438";tag=as0707d433
> To:
> Contact:
> Call-ID: 0f3bdcc9171ef53148e7bab413aea...@64.191.130.78
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Wed, 02 Sep 2009 21:10:39 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Type: application/sdp
> Content-Length: 412
> 
> v=0
> o=root 3831 3831 IN IP4 216.82.224.202
> s=session
> c=IN IP4 216.82.224.202
> t=0 0
> m=audio 17050 RTP/AVP 0 8 3 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> m=video 12426 RTP/AVP 31 34 103
> a=rtpmap:31 H261/9
> a=rtpmap:34 H263/9
> a=rtpmap:103 h263-1998/9
> a=sendrecv
> 

I know very little about how ringing works but are they providing any
kind of status information to you? Do you need to furnish the ring if
they are not? It seems to me I saw quite a few articles about providing
ring tone, what causes it to fail, and how to work around it.  I assume
you've searched for those already. Just a few thoughts - John
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com

http://www.spiritualoutreach.com
Making Christianity intelligible to secular society


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Re: [asterisk-users] outbound calls not ringing

[Ott Rose]

Thanks that is very helpful info. I am still trying to figure out how asterisk 
and freepbx works together. what do I add in those files to get the ringing to 
work. I checked teh Dail options under General Options and its set to tr.




> Date: Thu, 20 Aug 2009 10:51:25 +1200
> From: dun...@e-simple.co.nz
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] outbound calls not ringing
> 
> Generally with FreePBX the ring options are set in the General Options - 
> you can set the Dial options which are normally tr, but I guess that 
> isn't working for you.
> 
> The SIP files you could edit would have custom in their name, otherwise 
> your changes will be overwritten when you reload freepbx
> 
> You could put this in sip_general_custom.conf which will be included
> 
> Cheers Duncan
> 
> John A. Sullivan III wrote:
> > Oops! - You're using FreePBX - someone who knows more about FreePBX will
> > have to help you as I don't.  May I also suggest that you bottom post in
> > future responses rather than top post; that makes it a little easier to
> > follow.  Good luck - John
> >
> > On Wed, 2009-08-19 at 16:59 +, Ott Rose wrote:
> >   
> >> here is my sip.conf. i don't see it.
> >> ;;
> >> ; Do NOT edit this file as it is auto-generated by FreePBX. All
> >> modifications to ;
> >> ; this file must be done via the web gui. There are alternative files
> >> to make;
> >> ; custom modifications, details at:
> >> http://freepbx.org/configuration_files   ;
> >> ;;
> >> ;
> >>
> >> [general]
> >>
> >> ; These files will all be included in the [general] context
> >> ;
> >> #include sip_general_additional.conf
> >>
> >> ;sip_general_custom.conf is the proper file location for placing any
> >> sip general
> >> ;options that you might need set. For example: enable and force the
> >> sip jitterbuffer.
> >> ;If these settings are desired they should be set the
> >> sip_general_custom.conf file.
> >> ;
> >> ; jbenable=yes
> >> ; jbforce=yes
> >> ;
> >> ;It is also the proper place to add the lines needed for sip nat'ing
> >> when going
> >> ;through a firewall.  For nat'ing you'd need to add the following
> >> lines:
> >> ; nat=yes , externip= , localhost= , and optionally fromdomain= .
> >> ;
> >> #include sip_general_custom.conf
> >>
> >> ;sip_nat.conf is here for legacy support reasons and for those that
> >> upgrade
> >> ;from previous versions.  If you have this file with lines in it
> >> please make
> >> ;sure they are not duplicated in sip_general_custom.conf, if so remove
> >> them
> >> ;from sip_nat.conf as sip_general_custom.conf will have precedence.
> >> #include sip_nat.conf
> >>
> >> ;sip_registrations_custom.conf is for any customizations you might
> >> need to do to
> >> ;the automatically generated registrations that FreePBX makes.
> >> ;
> >> #include sip_registrations_custom.conf
> >> #include sip_registrations.conf
> >>
> >> ; These files should all be expected to come after the [general]
> >> context
> >> ;
> >> #include sip_custom.conf
> >> #include sip_additional.conf
> >>
> >> ;sip_custom_post.conf If you have extra parameters that are needed for
> >> a
> >> ;extension to work to for example, those go here.  So you have
> >> extension
> >> ;1000 defined in your system you start by creating a line [1000](+) in
> >> this
> >> ;file.  Then on the next line add the extra parameter that is needed.
> >> ;When the sip.conf is loaded it will append your additions to the end
> >> of
> >> ;that extension.
> >> ;
> >> #include sip_custom_post.conf
> >>
> >>
> >> 
> >>> From: jsulli...@opensourcedevel.com
> >>> To: asterisk-users@lists.digium.com
> >>> Date: Wed, 19 Aug 2009 12:17:15 -0400
> >>> Subject: Re: [asterisk-users] outbound calls not ringing
> >>>
> >>> sip.conf
> >>>
> >>> On Wed, 2009-08-19 at 15:55 +, Ott Rose wrote:
> >>>   
> >>>> we are using Aastra 57i
> >>>>
> >>>> i don't s

Re: [asterisk-users] outbound calls not ringing

[Duncan Turnbull]

Generally with FreePBX the ring options are set in the General Options - 
you can set the Dial options which are normally tr, but I guess that 
isn't working for you.

The SIP files you could edit would have custom in their name, otherwise 
your changes will be overwritten when you reload freepbx

You could put this in sip_general_custom.conf which will be included

Cheers Duncan

John A. Sullivan III wrote:
> Oops! - You're using FreePBX - someone who knows more about FreePBX will
> have to help you as I don't.  May I also suggest that you bottom post in
> future responses rather than top post; that makes it a little easier to
> follow.  Good luck - John
>
> On Wed, 2009-08-19 at 16:59 +, Ott Rose wrote:
>   
>> here is my sip.conf. i don't see it.
>> ;;
>> ; Do NOT edit this file as it is auto-generated by FreePBX. All
>> modifications to ;
>> ; this file must be done via the web gui. There are alternative files
>> to make;
>> ; custom modifications, details at:
>> http://freepbx.org/configuration_files   ;
>> ;;
>> ;
>>
>> [general]
>>
>> ; These files will all be included in the [general] context
>> ;
>> #include sip_general_additional.conf
>>
>> ;sip_general_custom.conf is the proper file location for placing any
>> sip general
>> ;options that you might need set. For example: enable and force the
>> sip jitterbuffer.
>> ;If these settings are desired they should be set the
>> sip_general_custom.conf file.
>> ;
>> ; jbenable=yes
>> ; jbforce=yes
>> ;
>> ;It is also the proper place to add the lines needed for sip nat'ing
>> when going
>> ;through a firewall.  For nat'ing you'd need to add the following
>> lines:
>> ; nat=yes , externip= , localhost= , and optionally fromdomain= .
>> ;
>> #include sip_general_custom.conf
>>
>> ;sip_nat.conf is here for legacy support reasons and for those that
>> upgrade
>> ;from previous versions.  If you have this file with lines in it
>> please make
>> ;sure they are not duplicated in sip_general_custom.conf, if so remove
>> them
>> ;from sip_nat.conf as sip_general_custom.conf will have precedence.
>> #include sip_nat.conf
>>
>> ;sip_registrations_custom.conf is for any customizations you might
>> need to do to
>> ;the automatically generated registrations that FreePBX makes.
>> ;
>> #include sip_registrations_custom.conf
>> #include sip_registrations.conf
>>
>> ; These files should all be expected to come after the [general]
>> context
>> ;
>> #include sip_custom.conf
>> #include sip_additional.conf
>>
>> ;sip_custom_post.conf If you have extra parameters that are needed for
>> a
>> ;extension to work to for example, those go here.  So you have
>> extension
>> ;1000 defined in your system you start by creating a line [1000](+) in
>> this
>> ;file.  Then on the next line add the extra parameter that is needed.
>> ;When the sip.conf is loaded it will append your additions to the end
>> of
>> ;that extension.
>> ;
>> #include sip_custom_post.conf
>>
>>
>> 
>>> From: jsulli...@opensourcedevel.com
>>> To: asterisk-users@lists.digium.com
>>> Date: Wed, 19 Aug 2009 12:17:15 -0400
>>> Subject: Re: [asterisk-users] outbound calls not ringing
>>>
>>> sip.conf
>>>
>>> On Wed, 2009-08-19 at 15:55 +, Ott Rose wrote:
>>>   
>>>> we are using Aastra 57i
>>>>
>>>> i don't see that setting. where is it at?
>>>>
>>>> 
>>>>> From: jsulli...@opensourcedevel.com
>>>>> To: asterisk-users@lists.digium.com
>>>>> Date: Wed, 19 Aug 2009 11:07:21 -0400
>>>>> Subject: Re: [asterisk-users] outbound calls not ringing
>>>>>
>>>>> On Wed, 2009-08-19 at 13:54 +, Ott Rose wrote:
>>>>>   
>>>>>> I put a post on here about my issues with outbound calls not
>>>>>> 
>>>> ringing
>>>> 
>>>>>> but i haven't resolved it. so i am trying again.
>>>>>>
>>>>>> When i dial any outside number i dont get a ring tone at all.
>>>>>> 
>> when
>> 
>>>> the
>>>> 
>&

Re: [asterisk-users] outbound calls not ringing

[John A. Sullivan III]

Oops! - You're using FreePBX - someone who knows more about FreePBX will
have to help you as I don't.  May I also suggest that you bottom post in
future responses rather than top post; that makes it a little easier to
follow.  Good luck - John

On Wed, 2009-08-19 at 16:59 +, Ott Rose wrote:
> here is my sip.conf. i don't see it.
> ;;
> ; Do NOT edit this file as it is auto-generated by FreePBX. All
> modifications to ;
> ; this file must be done via the web gui. There are alternative files
> to make;
> ; custom modifications, details at:
> http://freepbx.org/configuration_files   ;
> ;;
> ;
> 
> [general]
> 
> ; These files will all be included in the [general] context
> ;
> #include sip_general_additional.conf
> 
> ;sip_general_custom.conf is the proper file location for placing any
> sip general
> ;options that you might need set. For example: enable and force the
> sip jitterbuffer.
> ;If these settings are desired they should be set the
> sip_general_custom.conf file.
> ;
> ; jbenable=yes
> ; jbforce=yes
> ;
> ;It is also the proper place to add the lines needed for sip nat'ing
> when going
> ;through a firewall.  For nat'ing you'd need to add the following
> lines:
> ; nat=yes , externip= , localhost= , and optionally fromdomain= .
> ;
> #include sip_general_custom.conf
> 
> ;sip_nat.conf is here for legacy support reasons and for those that
> upgrade
> ;from previous versions.  If you have this file with lines in it
> please make
> ;sure they are not duplicated in sip_general_custom.conf, if so remove
> them
> ;from sip_nat.conf as sip_general_custom.conf will have precedence.
> #include sip_nat.conf
> 
> ;sip_registrations_custom.conf is for any customizations you might
> need to do to
> ;the automatically generated registrations that FreePBX makes.
> ;
> #include sip_registrations_custom.conf
> #include sip_registrations.conf
> 
> ; These files should all be expected to come after the [general]
> context
> ;
> #include sip_custom.conf
> #include sip_additional.conf
> 
> ;sip_custom_post.conf If you have extra parameters that are needed for
> a
> ;extension to work to for example, those go here.  So you have
> extension
> ;1000 defined in your system you start by creating a line [1000](+) in
> this
> ;file.  Then on the next line add the extra parameter that is needed.
> ;When the sip.conf is loaded it will append your additions to the end
> of
> ;that extension.
> ;
> #include sip_custom_post.conf
> 
> 
> > From: jsulli...@opensourcedevel.com
> > To: asterisk-users@lists.digium.com
> > Date: Wed, 19 Aug 2009 12:17:15 -0400
> > Subject: Re: [asterisk-users] outbound calls not ringing
> > 
> > sip.conf
> > 
> > On Wed, 2009-08-19 at 15:55 +0000, Ott Rose wrote:
> > > 
> > > we are using Aastra 57i
> > > 
> > > i don't see that setting. where is it at?
> > > 
> > > > From: jsulli...@opensourcedevel.com
> > > > To: asterisk-users@lists.digium.com
> > > > Date: Wed, 19 Aug 2009 11:07:21 -0400
> > > > Subject: Re: [asterisk-users] outbound calls not ringing
> > > > 
> > > > On Wed, 2009-08-19 at 13:54 +, Ott Rose wrote:
> > > > > I put a post on here about my issues with outbound calls not
> > > ringing
> > > > > but i haven't resolved it. so i am trying again.
> > > > > 
> > > > > When i dial any outside number i dont get a ring tone at all.
> when
> > > the
> > > > > person picks up and starts to talk i can hear them fine. it
> sounds
> > > > > great. How do I start to troubleshot this?
> > > > 
> > > > What type of phones are giving you the problem? If I recall
> > > correctly,
> > > > our SIP phones had this problem depending on how the destination
> > > handled
> > > > signaling. We resolved it by adding progressinband=no (as
> opposed to
> > > > the default never - at least I think it is the default) but this
> > > > produces the problem of duplicate ring tones at times. Hope this
> > > helps
> > > > - John
> > > > -- 
> > > > John A. Sullivan III
> > > > Open Source Development Corporation
> > > > +1 207-985-7880
> > > > jsulli...@opensourcedevel.com
> > > 

Re: [asterisk-users] outbound calls not ringing

[Ott Rose]

here is my sip.conf. i don't see it.
;;
; Do NOT edit this file as it is auto-generated by FreePBX. All modifications 
to ;
; this file must be done via the web gui. There are alternative files to make   
 ;
; custom modifications, details at: http://freepbx.org/configuration_files  
 ;
;;
;

[general]

; These files will all be included in the [general] context
;
#include sip_general_additional.conf

;sip_general_custom.conf is the proper file location for placing any sip general
;options that you might need set. For example: enable and force the sip 
jitterbuffer.
;If these settings are desired they should be set the sip_general_custom.conf 
file.
;
; jbenable=yes
; jbforce=yes
;
;It is also the proper place to add the lines needed for sip nat'ing when going
;through a firewall.  For nat'ing you'd need to add the following lines:
; nat=yes , externip= , localhost= , and optionally fromdomain= .
;
#include sip_general_custom.conf

;sip_nat.conf is here for legacy support reasons and for those that upgrade
;from previous versions.  If you have this file with lines in it please make
;sure they are not duplicated in sip_general_custom.conf, if so remove them
;from sip_nat.conf as sip_general_custom.conf will have precedence.
#include sip_nat.conf

;sip_registrations_custom.conf is for any customizations you might need to do to
;the automatically generated registrations that FreePBX makes.
;
#include sip_registrations_custom.conf
#include sip_registrations.conf

; These files should all be expected to come after the [general] context
;
#include sip_custom.conf
#include sip_additional.conf

;sip_custom_post.conf If you have extra parameters that are needed for a
;extension to work to for example, those go here.  So you have extension
;1000 defined in your system you start by creating a line [1000](+) in this
;file.  Then on the next line add the extra parameter that is needed.
;When the sip.conf is loaded it will append your additions to the end of
;that extension.
;
#include sip_custom_post.conf


> From: jsulli...@opensourcedevel.com
> To: asterisk-users@lists.digium.com
> Date: Wed, 19 Aug 2009 12:17:15 -0400
> Subject: Re: [asterisk-users] outbound calls not ringing
> 
> sip.conf
> 
> On Wed, 2009-08-19 at 15:55 +, Ott Rose wrote:
> > 
> > we are using Aastra 57i
> > 
> > i don't see that setting. where is it at?
> > 
> > > From: jsulli...@opensourcedevel.com
> > > To: asterisk-users@lists.digium.com
> > > Date: Wed, 19 Aug 2009 11:07:21 -0400
> > > Subject: Re: [asterisk-users] outbound calls not ringing
> > > 
> > > On Wed, 2009-08-19 at 13:54 +, Ott Rose wrote:
> > > > I put a post on here about my issues with outbound calls not
> > ringing
> > > > but i haven't resolved it. so i am trying again.
> > > > 
> > > > When i dial any outside number i dont get a ring tone at all. when
> > the
> > > > person picks up and starts to talk i can hear them fine. it sounds
> > > > great. How do I start to troubleshot this?
> > > 
> > > What type of phones are giving you the problem? If I recall
> > correctly,
> > > our SIP phones had this problem depending on how the destination
> > handled
> > > signaling. We resolved it by adding progressinband=no (as opposed to
> > > the default never - at least I think it is the default) but this
> > > produces the problem of duplicate ring tones at times. Hope this
> > helps
> > > - John
> > > -- 
> > > John A. Sullivan III
> > > Open Source Development Corporation
> > > +1 207-985-7880
> > > jsulli...@opensourcedevel.com
> > > 
> > > http://www.spiritualoutreach.com
> > > Making Christianity intelligible to secular society
> > > 
> > > 
> > > ___
> > > -- Bandwidth and Colocation Provided by http://www.api-digital.com
> > --
> > > 
> > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> > > Register Now: http://www.astricon.net
> > > 
> > > asterisk-users mailing list
> > > To UNSUBSCRIBE or update options visit:
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
> > 
> > __
> > Hotmail® is up to 70% faster. Now good news travels really fast. Try
> > it now.
> > ___
> > 

Re: [asterisk-users] outbound calls not ringing

[John A. Sullivan III]

sip.conf

On Wed, 2009-08-19 at 15:55 +, Ott Rose wrote:
> 
> we are using Aastra 57i
> 
> i don't see that setting. where is it at?
> 
> > From: jsulli...@opensourcedevel.com
> > To: asterisk-users@lists.digium.com
> > Date: Wed, 19 Aug 2009 11:07:21 -0400
> > Subject: Re: [asterisk-users] outbound calls not ringing
> > 
> > On Wed, 2009-08-19 at 13:54 +, Ott Rose wrote:
> > > I put a post on here about my issues with outbound calls not
> ringing
> > > but i haven't resolved it. so i am trying again.
> > > 
> > > When i dial any outside number i dont get a ring tone at all. when
> the
> > > person picks up and starts to talk i can hear them fine. it sounds
> > > great. How do I start to troubleshot this?
> > 
> > What type of phones are giving you the problem? If I recall
> correctly,
> > our SIP phones had this problem depending on how the destination
> handled
> > signaling. We resolved it by adding progressinband=no (as opposed to
> > the default never - at least I think it is the default) but this
> > produces the problem of duplicate ring tones at times. Hope this
> helps
> > - John
> > -- 
> > John A. Sullivan III
> > Open Source Development Corporation
> > +1 207-985-7880
> > jsulli...@opensourcedevel.com
> > 
> > http://www.spiritualoutreach.com
> > Making Christianity intelligible to secular society
> > 
> > 
> > ___
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com
> --
> > 
> > AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> > Register Now: http://www.astricon.net
> > 
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
> __
> Hotmail® is up to 70% faster. Now good news travels really fast. Try
> it now.
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> Register Now: http://www.astricon.net
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com

http://www.spiritualoutreach.com
Making Christianity intelligible to secular society


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Re: [asterisk-users] outbound calls not ringing

[Ott Rose]

we are using Aastra 57i

i don't see that setting. where is it at?

> From: jsulli...@opensourcedevel.com
> To: asterisk-users@lists.digium.com
> Date: Wed, 19 Aug 2009 11:07:21 -0400
> Subject: Re: [asterisk-users] outbound calls not ringing
> 
> On Wed, 2009-08-19 at 13:54 +, Ott Rose wrote:
> > I put a post on here about my issues with outbound calls not ringing
> > but i haven't resolved it. so i am trying again.
> > 
> > When i dial any outside number i dont get a ring tone at all. when the
> > person picks up and starts to talk i can hear them fine. it sounds
> > great. How do I start to troubleshot this?
> 
> What type of phones are giving you the problem? If I recall correctly,
> our SIP phones had this problem depending on how the destination handled
> signaling.  We resolved it by adding progressinband=no (as opposed to
> the default never - at least I think it is the default) but this
> produces the problem of duplicate ring tones at times.  Hope this helps
> - John
> -- 
> John A. Sullivan III
> Open Source Development Corporation
> +1 207-985-7880
> jsulli...@opensourcedevel.com
> 
> http://www.spiritualoutreach.com
> Making Christianity intelligible to secular society
> 
> 
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> Register Now: http://www.astricon.net
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

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Re: [asterisk-users] outbound calls not ringing

[John A. Sullivan III]

On Wed, 2009-08-19 at 13:54 +, Ott Rose wrote:
> I put a post on here about my issues with outbound calls not ringing
> but i haven't resolved it. so i am trying again.
> 
> When i dial any outside number i dont get a ring tone at all. when the
> person picks up and starts to talk i can hear them fine. it sounds
> great. How do I start to troubleshot this?

What type of phones are giving you the problem? If I recall correctly,
our SIP phones had this problem depending on how the destination handled
signaling.  We resolved it by adding progressinband=no (as opposed to
the default never - at least I think it is the default) but this
produces the problem of duplicate ring tones at times.  Hope this helps
- John
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com

http://www.spiritualoutreach.com
Making Christianity intelligible to secular society


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Re: [asterisk-users] outbound calls not ringing

[Danny Nicholas]

Have you tried putting a (,r) on your Dial command (dial
dahdi/1/18005551212,60,r) ?

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose
Sent: Wednesday, August 19, 2009 8:55 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] outbound calls not ringing

 

I put a post on here about my issues with outbound calls not ringing but i
haven't resolved it. so i am trying again.

When i dial any outside number i dont get a ring tone at all. when the
person picks up and starts to talk i can hear them fine. it sounds great.
How do I start to troubleshot this?

  _  

With Windows Live, you can organize, edit, and share your photos. Click
  here.

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Re: [asterisk-users] Outbound calls drop after 15 to 30 seconds.

[Guillaume Yziquel]

Steve Totaro a écrit :
> On Mon, Aug 3, 2009 at 8:20 AM, Guillaume Yziquel <
> guillaume.yziq...@citycable.ch> wrote:
> 
>> Hello.
>>
>> I've set up and configured an Asterisk server to make SIP phone calls to
>>  external classic phones.
>>
>> However, it happens that after 15 or 30 seconds, the phone call drops.
>> The SIP session still seems valid, but no sound comes through any more.
>>
>> How would you go through to troubleshoot this issue?
>>
>> All the best,
>>
>> Guillaume Yziquel.
>
> Make sure you have canreinvite set to no.

It was already set to 'no'

> Also, you may need to put an answer() in before your dial, I have dealt with
> that strangeness, call always drop at exactly 30 seconds.

Putting exten => _X.,n,Answer() in the dialplan doesn't change anything.

> That solution worked for me, but I could see how it could mess up CDRs and
> billing for some applications.

Maybe I'm having a different issue than you've been experiencing. What's 
rather painful is that nothing appears to show in the Asterisk CLI when 
this happens since it's obviously not a problem with the SIP connection.

How could I monitor the voice going in and out?

All the best,

Guillaume Yziquel.

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Re: [asterisk-users] Outbound calls drop after 15 to 30 seconds.

[Steve Totaro]

On Mon, Aug 3, 2009 at 8:20 AM, Guillaume Yziquel <
guillaume.yziq...@citycable.ch> wrote:

> Hello.
>
> I've set up and configured an Asterisk server to make SIP phone calls to
>  external classic phones.
>
> However, it happens that after 15 or 30 seconds, the phone call drops.
> The SIP session still seems valid, but no sound comes through any more.
>
> How would you go through to troubleshoot this issue?
>
> All the best,
>
> Guillaume Yziquel.
>
>
Make sure you have canreinvite set to no.

Also, you may need to put an answer() in before your dial, I have dealt with
that strangeness, call always drop at exactly 30 seconds.

That solution worked for me, but I could see how it could mess up CDRs and
billing for some applications.

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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Re: [asterisk-users] outbound calls not reaching vitelity

[Tom Poe]

John A. Sullivan III wrote:
> On Tue, 2009-07-28 at 13:33 -0500, Tom Poe wrote:
>   
>> Any vitelity customers with pbxinaflash boxes?  I'm able to call 
>> in-house, but failing to make outbound calls.  My assigned server at 
>> vitelity is not reachable.  I can ping to my ISP OK.
>> Any help appreciated.  Such as actually how to make email contact with 
>> support at vitelity.  They're not responding.
>> Thanks, Tom
>> 
> 
> I'm not using pbxinaflash but I am using Vitelity and have had no
> problems at all - in fact very happy with them.  They should have given
> you a management portal for your account probably portal.vitelity.net.
> In there, there is an option to open a trouble ticket.  Hope this helps
> - John
>   
Thanks, much.  Will do.
Tom

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Re: [asterisk-users] outbound calls not reaching vitelity

[John A. Sullivan III]

On Tue, 2009-07-28 at 13:33 -0500, Tom Poe wrote:
> Any vitelity customers with pbxinaflash boxes?  I'm able to call 
> in-house, but failing to make outbound calls.  My assigned server at 
> vitelity is not reachable.  I can ping to my ISP OK.
> Any help appreciated.  Such as actually how to make email contact with 
> support at vitelity.  They're not responding.
> Thanks, Tom

I'm not using pbxinaflash but I am using Vitelity and have had no
problems at all - in fact very happy with them.  They should have given
you a management portal for your account probably portal.vitelity.net.
In there, there is an option to open a trouble ticket.  Hope this helps
- John
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com

http://www.spiritualoutreach.com
Making Christianity intelligible to secular society


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Re: [Asterisk-Users] outbound calls to sip urls

[Jon-o Addleman]

On Mon, Apr 24, 2006 at 04:46:29PM +0530, Ajit spake thusly:
> Hi,
>  I wish to use the manager API to make an outbound call to a sip
> url,subsequently play a prompt and hangup.Any hints on how to acheive
> this/feasability will be much appreciated.

I'm no expert, but it looks simple enough to me - just use the originate
action to call with something like this:

Action: Originate
channel: SIP/[EMAIL PROTECTED]
context: testcontext
extension: extensiontosendtheprompt
priority: 1

So that extension will just send the prompt and then hang up.

-- 
Jon-o Addleman - http://redowl.dyndns.org
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Re: [Asterisk-Users] Outbound calls are failing

[nrbwpi]

Thanks

Inserting a "w" did resolve the problem.  I saw another post from
today where somebody else is having the same problem with a 
TDM2400P.  Hopefully someday Asterisk will be coded to wait for a dial tone.

nb


On 4/19/06, Time Bandit <[EMAIL PROTECTED]> wrote:
>  When dialing an outbound number, sometimes all the digits are not dialed> properly on the outside line. In the dial plan I added a SayDigits to the> outbound rule and it properly reads back the entire number that was entered
> on the phone before dialing.>>  Is asterisk dialing too quickly, is there anyway to insert a pause or wait> for a dial tone on the external line?* is probably starting to dial too fast. Try to add a w in your dial
string to make it wait.Like : Dial(ZAP/g0,w${EXTEN})w adds half a second pause. You can put more w to make it wait longer.hth___--Bandwidth and Colocation provided by 
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Re: [Asterisk-Users] Outbound calls are failing

[Time Bandit]

>  When dialing an outbound number, sometimes all the digits are not dialed
> properly on the outside line. In the dial plan I added a SayDigits to the
> outbound rule and it properly reads back the entire number that was entered
> on the phone before dialing.
>
>  Is asterisk dialing too quickly, is there anyway to insert a pause or wait
> for a dial tone on the external line?
* is probably starting to dial too fast. Try to add a w in your dial
string to make it wait.
Like : Dial(ZAP/g0,w${EXTEN})

w adds half a second pause. You can put more w to make it wait longer.

hth
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Re: [Asterisk-Users] Outbound calls through Broadvoice

[Ronald Wiplinger]

Mike Raley wrote:
Hi all, a noob here,  I am trying to get outbound calls through 
asterisk working with Broadvoice.


I have consulted the following two online tutorials:

http://www.broadvoice.com/support_install_asterisk.html

http://www.voip-info.org/wiki/view/Asterisk+settings+Broadvoice

in an effort to make outbound calls.
My current settings are as follows:

sip.conf

register => 
[EMAIL PROTECTED]::[EMAIL PROTECTED]/XX 



where XX = our phone number including area code
and  is our broadvoice defined secret

[sip.braodvoice.com]

maybe just a typo, br_OA_dvoice

I gone away from broadvoice, since they admitted to have troubles and I 
had still to pay for NO phone call !!! (multiple lines)



bye

Ronald Wiplinger

type=peer
dynamic=yes
username=XX
fromuser=XX
authname=XX
user=phone
secret=
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
outboundproxy=sip.broadvoice.com
insecure=very
dtmfmode=inband
dtmf=inband
canreinvite=no
context=incoming

I receive the following error through asterisk when attempting a call:

Apr  8 13:08:43 WARNING[17425]: chan_sip.c:9634 
handle_response_invite: Forbidden - wrong password on authentication 
for INVITE to '"My Name" ;tag=as23aa39db'


Now, we can receive incoming calls perfectly fine, but I just can't 
wrap my head around what is wrong with the outgoing.  I figure it's 
got to be the way I am passing the phone number to call to Broadvoice:


exten => _3XNXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten => _3XNXXNXX,2,Congestion

or possibly, the fact that my name is showing up in the outbound call, 
but the account isn't registered to my name, but someone else where I 
work.


or my conf files are wrong somehow?

Otherwise, I got nothing.

Any help would be greatly appreciated by one frustrated noob!

oh, please CC me at mraley [at] syndiogroup.com  [remove the 
 species]


Thanks!
Mike




--
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Re: [Asterisk-Users] Outbound calls unpredictable

[Matt Riddell]

Frank wrote:
Thats amazing! Worked like a charm...any explanations as to why this happens? 
Basically some connections require you to wait a little bit before 
dialling the number.  Without the w's it dials straight away.  With them 
it pauses and then dials.

On Wednesday 19 January 2005 03:21 am, Matt Riddell wrote:
Frank wrote:
I've been looking through the archives and have not been able to find
anyone with a similar problem but perhaps I'm not searching in the right
places. The problem is that my outbound call sometimes go though and
sometimes don't. If someone can point me in the right direction it will 
be highly appreciated.
You could try altering you dial line so that it starts with a few w's:
exten => _9X.,1,Dial(Zap/1/www${EXTEN:1})
--
Cheers,
Matt Riddell
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Re: [Asterisk-Users] Outbound calls unpredictable

[Frank]

Thats amazing! Worked like a charm...any explanations as to why this happens? 

On Wednesday 19 January 2005 03:21 am, Matt Riddell wrote:
> Frank wrote:
> > I've been looking through the archives and have not been able to find
> > anyone with a similar problem but perhaps I'm not searching in the right
> > places. The problem is that my outbound call sometimes go though and
> > sometimes don't. If someone can point me in the right direction it will 
> > be highly appreciated.
>
> You could try altering you dial line so that it starts with a few w's:
>
> exten => _9X.,1,Dial(Zap/1/www${EXTEN:1})
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Re: [Asterisk-Users] Outbound calls unpredictable

[Matt Riddell]

Frank wrote:
I've been looking through the archives and have not been able to find anyone 
with a similar problem but perhaps I'm not searching in the right places. The 
problem is that my outbound call sometimes go though and sometimes don't. If 
someone can point me in the right direction it will  be highly appreciated.
You could try altering you dial line so that it starts with a few w's:
exten => _9X.,1,Dial(Zap/1/www${EXTEN:1})
--
Cheers,
Matt Riddell
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Re: [Asterisk-Users] Outbound calls unpredictable

[Steven Critchfield]

On Tue, 2005-01-18 at 05:20 -0500, Frank wrote:
> I've been looking through the archives and have not been able to find anyone 
> with a similar problem but perhaps I'm not searching in the right places. The 
> problem is that my outbound call sometimes go though and sometimes don't. If 
> someone can point me in the right direction it will  be highly appreciated.

Care to give some logs or other information so we can think about
helping you?
-- 
Steven Critchfield <[EMAIL PROTECTED]>

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