Re: [asterisk-users] SIP call drops after 32 seconds, but only when....
Hi Yves.. This may be silly... but what is the useragent of your sip configuration? In the case that useragent has some special characters like (., please remove it and tell us if there is any change!!. Regards. rv 2014-11-22 14:50 GMT-03:00 Eric Wieling ewiel...@nyigc.com: Try setting directmedia=no in sip.conf. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Yves A. Sent: Saturday, November 22, 2014 8:06 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] SIP call drops after 32 seconds, but only when Am 22.11.2014 um 12:51 schrieb Andreas Sikkema: but as soon as I configure another sip registration on another server, outgoing calls drop after 32 seconds. Are both your servers behind the same NAT router? thanks for taking part... I don´t know... one is siptrunk.ovh.net and the other one is sip.ovh.fr how can i determine and how could that affect... I mean... why do they interfere at all? thanks, yves --- Diese E-Mail wurde von Avast Antivirus-Software auf Viren geprüft. http://www.avast.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] agi get_data noanswer
Eric is correct. There is no way to send dtmf while the call has not been
answered.
But us very confusing the read command, in specific option = n(noanswer)
to read digits even if the line is not up
My AGI line is the following
$AGI-exec(READ,umenu,VARXX,1,n,2,7);
The command works, but there is no dtmf negotiation
$AGI-exec(READ,umenu,VARXX,1,,2,7);
The command works, but there is a kind of answer
What is the purpose of this noanswer option in a read command when it is
imposible to read?.
Is there any way to negotiate with the end user in this early media
situation?
Thanks in advance.
rv
2014-08-07 20:02 GMT-04:00 Eric Wieling ewiel...@nyigc.com:
Generally the only thing you are allowed to do before answer is send
audio. You can’t receive audio and can’t receive DTMF. I assume it is to
prevent people from doing exactly what you are trying to do --- trying to
have two way communications without paying for the call.
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rafael Visser
*Sent:* Thursday, August 07, 2014 4:56 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] agi get_data noanswer
Hi Guys..
I am making an anoucement machine that is not allowed to answer the call
due to a billing issue.
I found that Playback with noanwser is usefull in this case.
$AGI-exec('Playback',$message,noanswer)}
But when i request some values to the user with get_data, i think there is
an answer anywere.
Is there a way to get_data without answering the call?
Thanks in advance!!
rv
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Re: [asterisk-users] agi get_data noanswer
I am talking about sip on asterisk 11.10.2
rv
2014-08-12 19:28 GMT-04:00 Eric Wieling ewiel...@nyigc.com:
I do not know, maybe some of the other channel drivers sccp or sip support
it.
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rafael Visser
*Sent:* Tuesday, August 12, 2014 7:24 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] agi get_data noanswer
Eric is correct. There is no way to send dtmf while the call has not been
answered.
But us very confusing the read command, in specific option = n(noanswer)
to read digits even if the line is not up
My AGI line is the following
$AGI-exec(READ,umenu,VARXX,1,n,2,7);
The command works, but there is no dtmf negotiation
$AGI-exec(READ,umenu,VARXX,1,,2,7);
The command works, but there is a kind of answer
What is the purpose of this noanswer option in a read command when it is
imposible to read?.
Is there any way to negotiate with the end user in this early media
situation?
Thanks in advance.
rv
2014-08-07 20:02 GMT-04:00 Eric Wieling ewiel...@nyigc.com:
Generally the only thing you are allowed to do before answer is send
audio. You can’t receive audio and can’t receive DTMF. I assume it is to
prevent people from doing exactly what you are trying to do --- trying to
have two way communications without paying for the call.
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rafael Visser
*Sent:* Thursday, August 07, 2014 4:56 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] agi get_data noanswer
Hi Guys..
I am making an anoucement machine that is not allowed to answer the call
due to a billing issue.
I found that Playback with noanwser is usefull in this case.
$AGI-exec('Playback',$message,noanswer)}
But when i request some values to the user with get_data, i think there is
an answer anywere.
Is there a way to get_data without answering the call?
Thanks in advance!!
rv
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[asterisk-users] agi get_data noanswer
Hi Guys..
I am making an anoucement machine that is not allowed to answer the call
due to a billing issue.
I found that Playback with noanwser is usefull in this case.
$AGI-exec('Playback',$message,noanswer)}
But when i request some values to the user with get_data, i think there is
an answer anywere.
Is there a way to get_data without answering the call?
Thanks in advance!!
rv
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Re: [asterisk-users] agi get_data noanswer
Hi John.
I am making an inteligent annoucement resouce for a big ericsson switch. Is
just an ivr with agi applications.
The tricky thing try to make asterisk not to send answer. The perl
application with agi commands must be executed with out answering.
Something like
exten = 6009,1,Progress()
exten = 6009,n,Set(__INICIA=${EPOCH})
exten = 6009,n,Set(CHANNEL(language)=sc)
exten = 6009,n,AGI(anouncement.pl)
exten = 6009,n,Hangup()
Thanks anyway.
rv
2014-08-07 17:11 GMT-04:00 Tech Support aster...@voipbusiness.us:
What you may want to check out is the PlayTones and Ringing
applications in your dial plan. Asterisk will answer the call, but your
users won't know that because all they hear is the call still ringing.
After a certain amount of time passes, you can send them directly to
voicemail, hangup, run your scripts, or anything else you want to do with
the call. My dial plan snippet looks like this. Just an option.
exten = s,n(ringing),Answer
exten = s,n,PlayTones(ring)
exten = s,n,Ringing
exten = s,n,Wait(${TIMEOUT})
exten = s,n,GotoIf($[${BLOCKDEST} = 3]?s-NA-VOICEMAIL,1)
exten = s,n,GotoIf($[${CUSTCALLBLOCKACTION} = 3]?s-NA-VOICEMAIL,1) ;
exten = s,n,PlayTones(congestion)
exten = s,n,Congestion(10)
exten = s,n,Hangup
Regards;
John V.
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rafael Visser
*Sent:* Thursday, August 07, 2014 4:56 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] agi get_data noanswer
Hi Guys..
I am making an anoucement machine that is not allowed to answer the call
due to a billing issue.
I found that Playback with noanwser is usefull in this case.
$AGI-exec('Playback',$message,noanswer)}
But when i request some values to the user with get_data, i think there is
an answer anywere.
Is there a way to get_data without answering the call?
Thanks in advance!!
rv
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[asterisk-users] dialplan =how many concurrent calls
Hi guys. Does somebody knows how to get the concurrent calls from the dial plan? Or. How can i control not to run more than n simultaneus agi applications? Thanks in advance. rv -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dialplan =how many concurrent calls
Works fine..
Thanks Asghar!
rv
2014-07-10 9:35 GMT-04:00 Asghar Mohammad asghar...@gmail.com:
you can use GROUP and GROUP_COUNT
n,Set(GROUP()=aname)
n,GotoIf($[${GROUP_COUNT(aname)} 8]?${EXTEN},200)
200,Hangup
On Thu, Jul 10, 2014 at 3:24 PM, Rafael Visser visser.raf...@gmail.com
wrote:
Hi guys.
Does somebody knows how to get the concurrent calls from the dial plan?
Or.
How can i control not to run more than n simultaneus agi applications?
Thanks in advance.
rv
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Re: [asterisk-users] OPTIONS Request without username - Forbidden
So SIP/2.0 403 Forbidden is a valid response for qualify purpose Thanks Brian!! rv 2014-07-03 5:18 GMT-04:00 Brian LaVallee b.laval...@globaltank.jp: Hi Rafael, It's nothing to worry about -and- you might not be able to fix it. But it's nothing to worry about. -- Asterisk is using OPTIONS like a ping, qualify=yes. Since 403 is a *valid* SIP reply, the remote SIP service is considered reachable. My carrier replies with 405 Method Not Allowed, but it still indicates the SIP connection is up and working. -- Some carriers do not support OPTIONS. This is normally due to a proxy or other security mechanisms. Remember, OPTIONS is a request for what commands will be accepted. Sometime, you just don't want to advertise that kind of information. -- Check an INBOUND call (INVITE) and it will typically show what the carrier allows. If OPTIONS is not listed, there's nothing you can do. IP CARRIER_IP.sip LOCAL_IP.sip: UDP, length 870 E.@.9.9:=...j.p.n$BINVITE sip:212555@LOCAL_IP:5060 SIP/2.0 Via: SIP/2.0/UDP CARRIER_IP:5060;branch=z9hG4bKdac2492a2a1a086867cfb73fb2b5c8ac Via: SIP/2.0/UDP PROXY_IP:5060;branch=z9hG4bK09B55db052ffec696bd From: sip:212555@PROXY_IP:5060;tag=gK094dc1e4 To: sip:212555@CARRIER_IP:5060;tag=as2953dd14 Call-ID: 1980326667_35899190@PROXY_IP CSeq: 7852 INVITE Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,REGISTER,PRACK,UPDATE snip Accept: application/sdp Sincerely, Brian LaVallee On 6/25/14, 11:30 PM, Rafael Visser wrote: Hi gurus!!! I have a Freepbx with Asterisk 1.8.25.0 with a sip trunk on the pstn Every minute asterisk sends an OPTION Request, i beleived that it's related to qualify functions. The every minute annoyng answer of the pstn is 403 Forbidden. Some people told that asterisk is not sending the username in the OPTION, required by the pstn. Taking a look of the example of rfc3261.txt (pg 67), we found carol, so it makingme see that i am missing some config. OPTIONS sip:ca...@chicago.com SIP/2.0 Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKhjhs8ass877 Max-Forwards: 70 To: sip:ca...@chicago.com Is it wright? How can i instruct FREEPBX to send the username in the option request? Sorry for this silly question but a found no answer googling. Thans in advance. rv This is the debug of the case Reliably Transmitting (NAT) to 201.217.31.XX:5060: OPTIONS sip:201.217.31.10 SIP/2.0 Via: SIP/2.0/UDP 18x.16.204.XXX:6060;branch=z9hG4bK1d8715df;rport Max-Forwards: 70 From: Unknown sip:59x212376...@186.16.204.xxx:6060;tag=as4491c6af To: sip:201.217.31.10 Contact: sip:59x212376...@18x.16.204.xxx:6060 Call-ID: 4f02699e2632410c359e1ee43a021...@186.16.204.xxx:6060 CSeq: 102 OPTIONS User-Agent: FPBX-2.11.0(1.8.25.0) Date: Wed, 25 Jun 2014 13:47:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- SIP read from UDP:201.217.31.XX:5060 --- SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 18x.16.204.XXX:6060;received=18x.16.204.XXX;branch=z9hG4bK1d8715df;rport=5060 From: Unknown sip:59x212376...@18x.16.204.xxx:6060;tag=as4491c6af To: sip:201.217.31.XX;tag=aprqngfrt-nm50ea1c6 Call-ID: 4f02699e2632410c359e1ee43a021...@18x.16.204.xxx:6060 CSeq: 102 OPTIONS This is the peer. * Name : desde-XopaXo-2376XXX Secret : Set MD5Secret: Not set Remote Secret: Not set Context : from-trunk Subscr.Cont. : Not set Language : AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : MOH Suggest : Mailbox : VM Extension : *97 LastMsgsSent : 32767/65535 Call limit : 0 Max forwards : 0 Dynamic : No Callerid : MaxCallBR: 384 kbps Expire : -1 Insecure : port,invite Force rport : Yes ACL : No DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: -1 DirectMedia : No PromiscRedir : No User=Phone : No Video Support: No Text Support : No Ign SDP ver : No Trust RPID : No Send RPID: No Subscriptions: Yes Overlap dial : Yes DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : 201.217.31.10 Addr-IP : 201.217.31.10:5060 Defaddr-IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: 595212376458 SIP Options : timer Codecs : 0xe (gsm|ulaw|alaw) Codec Order : (ulaw:20,alaw:20,gsm:20) Auto-Framing : No Status : OK (36 ms) Useragent: Reg. Contact : Qualify Freq : 6 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk
Re: [asterisk-users] AGI script VERBOSE cmd
what if yoy change the verbose on the cli? cli core set verbose 4 and then try again i usually put on my perl agi something like $verbose=5; AGI-verbose(the number is $number, $verbose); hope it helps. rv 2014-06-27 11:24 GMT-04:00 Bryant Zimmerman brya...@zktech.com: I am working on an AGI script and all is going well except I can not get any of my VERBOSE commands to display. Is there any undocumented reason for this to occur? I am able to set variables, call other commands ect.. I am sending my verbose command in the following format (VERBOSE Message to send 4) Any ideas what I might be doing incorrect? Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Executing an AGI python script in Asterisk after call is bridged.
Hi Anurag.
I didn't undertand much you question. But you have a dial option to a
macro when b answers
example...
exten =
_+2XX!,n,Dial(SIP/sip.XX/+${DESTINO},20,rgM(acceptcall^${SESSIONID})S(${MAXCALLTIME}))
[macro-acceptcall]
; this macro is executed when b answers, requesting b if is interested to
pay the bill
exten = s,1,AGI(your-agi-program.pl)
exten = s,2,others...
Regards..
rv
2014-06-26 11:19 GMT-04:00 Anurag Rana anuragrana31...@gmail.com:
Hi All,
There is an option of starting the recording of call after the call is
bridged. [ b option].
Is there any way of running an AGI script only if call is bridged
otherwise not.
Thanks
--
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http://newbie42.blogspot.in/
On the trampoline of life's experiences, Striving towards a saintly life
in the midst of these materialistic turbulences.
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Re: [asterisk-users] Executing an AGI python script in Asterisk after call is bridged.
Ok.
in this link you will find some easy macro
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
Explanation
exten = _+2XX!,n,Dial(SIP/sip.XX/+${DESTINATION},20,rgM(acceptcall^${
SESSIONID})S(${MAXCALLTIME}))
Dial comand
to number DESTINATION
with timout of 20 seconds
r=ring sound
g= when hungs, continue with the dialplan (keep alive de application)
M=execute macro acceptcall passing the value SESSIONID
acceptcall=name of the macro
MAXCALLTIME is a value in second that the call is allowed to (prepaid)
So maybe you have to think somethin like this
exten=,1,Dial(SIP/,,rgM(mymacro))
and at the end of your dialplan
[macro-mymacro]
exten=s,1,AGI ( pythonscript.py )
It's not easy...
Good Luck.
2014-06-26 13:57 GMT-04:00 Anurag Rana anuragrana31...@gmail.com:
Thanks Rafeal. This is what I needed.
But first line i.e.
exten = _+2XX!,n,Dial(SIP/sip.XX/+${DESTINO},20,rgM(acceptcall^${
SESSIONID})S(${MAXCALLTIME}))
is very complicated.
I have very simple plan which is as below.
[context-demo]
exten=,1,AGI ( pythonscript.py )
exten=,1,Dial(SIP/)
that all.
Now can you please explain me in simpler form.
I am sorry. I am a newbie.
On Thu, Jun 26, 2014 at 11:12 PM, Rafael Visser visser.raf...@gmail.com
wrote:
Hi Anurag.
I didn't undertand much you question. But you have a dial option to a
macro when b answers
example...
exten =
_+2XX!,n,Dial(SIP/sip.XX/+${DESTINO},20,rgM(acceptcall^${SESSIONID})S(${MAXCALLTIME}))
[macro-acceptcall]
; this macro is executed when b answers, requesting b if is interested
to pay the bill
exten = s,1,AGI(your-agi-program.pl)
exten = s,2,others...
Regards..
rv
2014-06-26 11:19 GMT-04:00 Anurag Rana anuragrana31...@gmail.com:
Hi All,
There is an option of starting the recording of call after the call is
bridged. [ b option].
Is there any way of running an AGI script only if call is bridged
otherwise not.
Thanks
--
Anurag Rana
http://newbie42.blogspot.in/
On the trampoline of life's experiences, Striving towards a saintly life
in the midst of these materialistic turbulences.
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[asterisk-users] OPTIONS Request without username - Forbidden
Hi gurus!!! I have a Freepbx with Asterisk 1.8.25.0 with a sip trunk on the pstn Every minute asterisk sends an OPTION Request, i beleived that it's related to qualify functions. The every minute annoyng answer of the pstn is 403 Forbidden. Some people told that asterisk is not sending the username in the OPTION, required by the pstn. Taking a look of the example of rfc3261.txt (pg 67), we found carol, so it makingme see that i am missing some config. OPTIONS sip:ca...@chicago.com SIP/2.0 Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKhjhs8ass877 Max-Forwards: 70 To: sip:ca...@chicago.com Is it wright? How can i instruct FREEPBX to send the username in the option request? Sorry for this silly question but a found no answer googling. Thans in advance. rv This is the debug of the case Reliably Transmitting (NAT) to 201.217.31.XX:5060: OPTIONS sip:201.217.31.10 SIP/2.0 Via: SIP/2.0/UDP 18x.16.204.XXX:6060;branch=z9hG4bK1d8715df;rport Max-Forwards: 70 From: Unknown sip:59x212376...@186.16.204.xxx:6060;tag=as4491c6af To: sip:201.217.31.10 Contact: sip:59x212376...@18x.16.204.xxx:6060 Call-ID: 4f02699e2632410c359e1ee43a021...@186.16.204.xxx:6060 CSeq: 102 OPTIONS User-Agent: FPBX-2.11.0(1.8.25.0) Date: Wed, 25 Jun 2014 13:47:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- SIP read from UDP:201.217.31.XX:5060 --- SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 18x.16.204.XXX:6060;received=18x.16.204.XXX;branch=z9hG4bK1d8715df;rport=5060 From: Unknown sip:59x212376...@18x.16.204.xxx:6060;tag=as4491c6af To: sip:201.217.31.XX;tag=aprqngfrt-nm50ea1c6 Call-ID: 4f02699e2632410c359e1ee43a021...@18x.16.204.xxx:6060 CSeq: 102 OPTIONS This is the peer. * Name : desde-XopaXo-2376XXX Secret : Set MD5Secret: Not set Remote Secret: Not set Context : from-trunk Subscr.Cont. : Not set Language : AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : MOH Suggest : Mailbox : VM Extension : *97 LastMsgsSent : 32767/65535 Call limit : 0 Max forwards : 0 Dynamic : No Callerid : MaxCallBR: 384 kbps Expire : -1 Insecure : port,invite Force rport : Yes ACL : No DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: -1 DirectMedia : No PromiscRedir : No User=Phone : No Video Support: No Text Support : No Ign SDP ver : No Trust RPID : No Send RPID: No Subscriptions: Yes Overlap dial : Yes DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : 201.217.31.10 Addr-IP : 201.217.31.10:5060 Defaddr-IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: 595212376458 SIP Options : timer Codecs : 0xe (gsm|ulaw|alaw) Codec Order : (ulaw:20,alaw:20,gsm:20) Auto-Framing : No Status : OK (36 ms) Useragent: Reg. Contact : Qualify Freq : 6 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No * Name : desde-XopaXo-2376XXX Secret : Set MD5Secret: Not set Remote Secret: Not set Context : from-trunk Subscr.Cont. : Not set Language : AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : MOH Suggest : Mailbox : VM Extension : *97 LastMsgsSent : 32767/65535 Call limit : 0 Max forwards : 0 Dynamic : No Callerid : MaxCallBR: 384 kbps Expire : -1 Insecure : port,invite Force rport : Yes ACL : No DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: -1 DirectMedia : No PromiscRedir : No User=Phone : No Video Support: No Text Support : No Ign SDP ver : No Trust RPID : No Send RPID: No Subscriptions: Yes Overlap dial : Yes DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : 201.217.31.XX Addr-IP : 201.217.31.XX:5060 Defaddr-IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: 59X212376XXX SIP Options : timer Codecs : 0xe (gsm|ulaw|alaw) Codec Order : (ulaw:20,alaw:20,gsm:20) Auto-Framing : No Status : OK (36 ms) Useragent: Reg. Contact : Qualify Freq : 6 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:
[asterisk-users] queues show some agents (In use) from the start
Hi gurus Some of my agents are in use with no call involved. Members: Local/0971230031@internal/n (In use) has taken no calls yet Local/0972105500@internal/n (In use) has taken no calls yet No Callers Is there a workarround so solve this? Wham am i doing wrong? If i change agent Local/0971230031 for an other mobile line works fine :( This is my config Asterisk 1.6.1.6 DAHDI Version: 2.6.1 libss7 version: 1.0.2 queues.conf [RCEN] musicclass = default strategy = rrmemory weight=0 wrapuptime=15 autopause=no setinterfacevar=yes setqueueentryvar=yes setqueuevar=yes eventwhencalled = yes ringinuse = no joinempty=yes member = Local/0972105500@internal/n member = Local/0971230031@internal/n Thanks in advance. rv -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk and ericsson apg212-60 caller id issue
Hi gurus: I have an asterisk workging fine with an ericsson apg212-60, the thing is that when asterisk dials to the ericsson the callerid is not shown on ericsson's network. The oposite works!! Do you have any idea to solve this issue? Thanks rv -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] the lenght of the uri affects on dialplan?
Hi Gurus.. I use asterisk for just for ivr. My issue is that when the switch changes it's host name from MSSASU1.MYDOMAIN to MSSSASU1.MYDOMAiN.COM or MSSSASU1.MYDOMAiN.COM.PY the call is rejected with No matching peer and the handle_request_invite: Sending fake auth rejection for device x. It doesn't match it's own default context. Also, it has somethig to do with the numbers of digits of the dialed number. Few digits works ok, 14 to more works wrong. Do you know what am i missing? Thanks in advance. Debug with long hostname (B is considered as an '*') --- SIP read from TCP:10.146.9.70:6240 --- INVITE sip:b56510123456789012...@sisivr03.mydomain.com.py;user=phone SIP/2.0 From: sip:971200...@mssasu1.mydomain.com.py;user=phone;tag=3016589695 To: sip:b56510123456789012...@sisivr03.mydomain.com.py;user=phone Max-Forwards: 70 Via: SIP/2.0/TCP MSSASU1.MYDOMAIN.COM.PY:5060;branch=z9hG4bK0035391821780096 Call-ID: 9cax8060616182201-bo...@mssasu1.mydomain.com.py CSeq: 7313 INVITE P-Asserted-Identity: sip:971200...@mssasu1.mydomain.com.py;user=phone Accept: application/sdp Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,PRACK,UPDATE P-Charging-Vector: icid-value=6510081000-0826-16155907;icid-generated-at=MSSASU1.MYDOMAIN.COM.PY;orig-ioi=MSSASU1.MYDOMAIN.COM.PY Supported: 100rel Content-Type: application/sdp Contact: sip:MSSASU1.MYDOMAIN.COM.PY:5060;transport=UDP Content-Length: 414 v=0 o=- 7530078 7530078 IN IP4 MSSASU1.MYDOMAIN.COM.PY s=- t=0 0 a=sendrecv m=audio 13802 RTP/AVP 8 96 18 97 c=IN IP4 10.143.1.67 b=RR:0 b=RS:0 a=rtpmap:8 PCMA/8000 a=rtpmap:96 AMR/8000 a=fmtp:96 mode-set=0,2,4,7;mode-change-period=2;mode-change-capability=2;mode-change-neighbor=1;max-red=0 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-15 a=maxptime:40 - --- (15 headers 17 lines) --- Sending to 10.146.9.70:5060 (no NAT) Using INVITE request as basis request - 9cax8060616182201-bo...@mssasu1.mydomain.com.py No matching peer for '971200152' from '10.146.9.70:6240' [Aug 26 16:15:55] NOTICE[6873]: chan_sip.c:21975 handle_request_invite: Sending fake auth rej ection for device sip:971200...@mssasu1.mydomain.com.py;user=phone;tag=3016589695 # --- Reliably Transmitting (no NAT) to 10.146.9.70:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/TCP MSSASU1.MYDOMAIN.COM.PY:5060;branch=z9hG4bK0035391821780096;received=10.146.9.70 From: sip:971200...@mssasu1.mydomain.com.py;user=phone;tag=3016589695 To: sip:b56510123456789012...@sisivr03.mydomain.com.py;user=phone;tag=as4cfd0d54 Call-ID: 9cax8060616182201-bo...@mssasu1.mydomain.com.py CSeq: 7313 INVITE Server: Asterisk PBX 1.8.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=35ff0feb Content-Length: 0 Short hostname on switch === Connected to Asterisk 1.8.7.0 currently running on fdosis-ims1 (pid = 25430) fdosis-ims1*CLI core set verbose 1 Verbosity was 0 and is now 1 --- SIP read from UDP:10.146.9.70:5060 --- INVITE sip:b56510123456789012...@sisivr03.mydomain.com.py;user=phone SIP/2.0 From: sip:971200152@MSSASU1.MYDOMAIN;user=phone;tag=0046120455 To: sip:b56510123456789012...@sisivr03.mydomain.com.py;user=phone Max-Forwards: 70 Via: SIP/2.0/UDP MSSASU1.MYDOMAIN:5060;branch=z9hG4bK0038670956791982 Call-ID: qDaQ1240646182201-AKDE-@MSSASU1.MYDOMAIN CSeq: 14481 INVITE P-Asserted-Identity: sip:971200152@MSSASU1.MYDOMAIN;user=phone Accept: application/sdp llow: INVITE,ACK,OPTIONS,BYE,CANCEL,PRACK,UPDATE P-Charging-Vector: icid-value=A6D6B81000-0826-16440101;icid-generated-at=MSSASU1.MYDOMAIN;orig-ioi=MSSASU1.MYDOMAIN.COM.PY Supported: 100rel Content-Type: application/sdp Contact: sip:MSSASU1.MYDOMAIN:5060;transport=UDP Content-Length: 407 v=0 o=- 8986991 8986991 IN IP4 MSSASU1.MYDOMAIN s=- t=0 0 a=sendrecv m=audio 30838 RTP/AVP 8 96 18 97 c=IN IP4 10.143.1.68 b=RR:0 b=RS:0 a=rtpmap:8 PCMA/8000 a=rtpmap:96 AMR/8000 a=fmtp:96 mode-set=0,2,4,7;mode-change-period=2;mode-change-capability=2;mode-change-neighbor=1;max-red=0 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-15 a=maxptime:40 - --- (15 headers 17 lines) --- Sending to 10.146.9.70:5060 (no NAT) Using INVITE request as basis request - qDaQ1240646182201-AKDE-@MSSASU1.MYDOMAIN Found peer 'sip.ericsson' for '971200152' from 10.146.9.70:5060 Found RTP audio format 8 Found RTP audio format 96 Found RTP audio format 18 Found RTP audio format 97 Found audio description format PCMA for ID 8 Found unknown media description format AMR for ID 96 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 97 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x108 (alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf):
Re: [asterisk-users] the lenght of the uri affects on dialplan?
Ok... sip.conf [general] context=default ; Default context for incoming calls allowguest=no ; Allow or reject guest calls -sin password- (default is yes) allowoverlap=no ; Disable overlap dialing support. (Default is yes) udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all) tcpenable=yes; Enable server for incoming TCP connections (default is no) tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces) srvlookup=yes ; Enable DNS SRV lookups on outbound calls relaxdtmf=yes dtmfmode=inband ;rfc2833compensate=yes users.conf [general] fullname = New User userbase = 6000 hasvoicemail = yes vmsecret = 1234 hassip = yes hasiax = no hash323 = no hasmanager = no callwaiting = yes threewaycalling = yes callwaitingcallerid = yes transfer = yes canpark = yes cancallforward = yes callreturn = yes callgroup = 1 pickupgroup = 1 allowguest=no ; Allow or reject guest calls -sin password- (default is yes) [sip.ericsson] ;cambios allowguest hosts ;allowguest=no ; Allow or reject guest calls -sin password- (default is yes) type=friend calllimit=200 fromuser=ivr1 dtmfmode=inband username=administrador context=incoming-sip-ericsson host=10.146.9.70 host=ericsson host=MSSASU1.MYDOMAIN.COM.PY port=5060 disallow=all allow=alaw allow=gsm allow=ulaw qualify=yes insecure=no Date: Mon, 27 Aug 2012 03:42:51 +0500 From: fai...@vopium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] the lenght of the uri affects on dialplan? mention the complete scnario and your sip.conf. Regards, Faisal (sent from phone) Rafael Visser rafael_vis...@hotmail.com wrote: Hi Gurus.. I use asterisk for just for ivr. My issue is that when the switch changes it's host name from MSSASU1.MYDOMAIN to MSSSASU1.MYDOMAiN.COM or MSSSASU1.MYDOMAiN.COM.PY the call is rejected with No matching peer and the handle_request_invite: Sending fake auth rejection for device x. It doesn't match it's own default context. Also, it has somethig to do with the numbers of digits of the dialed number. Few digits works ok, 14 to more works wrong. Do you know what am i missing? Thanks in advance. Debug with long hostname (B is considered as an '*') --- SIP read from TCP:10.146.9.70:6240 --- INVITE sip:b56510123456789012...@sisivr03.mydomain.com.py;user=phone SIP/2.0 From: sip:971200...@mssasu1.mydomain.com.py;user=phone;tag=3016589695 To: sip:b56510123456789012...@sisivr03.mydomain.com.py;user=phone Max-Forwards: 70 Via: SIP/2.0/TCP MSSASU1.MYDOMAIN.COM.PY:5060;branch=z9hG4bK0035391821780096 Call-ID: 9cax8060616182201-bo...@mssasu1.mydomain.com.py CSeq: 7313 INVITE P-Asserted-Identity: sip:971200...@mssasu1.mydomain.com.py;user=phone Accept: application/sdp Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,PRACK,UPDATE P-Charging-Vector: icid-value=6510081000-0826-16155907;icid-generated-at=MSSASU1.MYDOMAIN.COM.PY;orig-ioi=MSSASU1.MYDOMAIN.COM.PY Supported: 100rel Content-Type: application/sdp Contact: sip:MSSASU1.MYDOMAIN.COM.PY:5060;transport=UDP Content-Length: 414 v=0 o=- 7530078 7530078 IN IP4 MSSASU1.MYDOMAIN.COM.PY s=- t=0 0 a=sendrecv m=audio 13802 RTP/AVP 8 96 18 97 c=IN IP4 10.143.1.67 b=RR:0 b=RS:0 a=rtpmap:8 PCMA/8000 a=rtpmap:96 AMR/8000 a=fmtp:96 mode-set=0,2,4,7;mode-change-period=2;mode-change-capability=2;mode-change-neighbor=1;max-red=0 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-15 a=maxptime:40 - --- (15 headers 17 lines) --- Sending to 10.146.9.70:5060 (no NAT) Using INVITE request as basis request - 9cax8060616182201-bo...@mssasu1.mydomain.com.py No matching peer for '971200152' from '10.146.9.70:6240' [Aug 26 16:15:55] NOTICE[6873]: chan_sip.c:21975 handle_request_invite: Sending fake auth rej ection for device sip:971200...@mssasu1.mydomain.com.py;user=phone;tag=3016589695 # --- Reliably Transmitting (no NAT) to 10.146.9.70:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/TCP MSSASU1.MYDOMAIN.COM.PY:5060;branch=z9hG4bK0035391821780096;received=10.146.9.70 From: sip:971200...@mssasu1.mydomain.com.py;user=phone;tag=3016589695 To: sip:b56510123456789012...@sisivr03.mydomain.com.py;user=phone;tag=as4cfd0d54 Call-ID: 9cax8060616182201-bo...@mssasu1.mydomain.com.py CSeq: 7313 INVITE Server: Asterisk PBX 1.8.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=35ff0feb Content-Length: 0 Short hostname on switch === Connected to Asterisk 1.8.7.0 currently running on fdosis-ims1 (pid = 25430) fdosis-ims1*CLI core set verbose 1 Verbosity was 0 and is now 1
Re: [asterisk-users] the lenght of the uri affects on dialplan?
Sorry, the last config was not clear. I replaced for the following sip.conf [general] context=default ; Default context for incoming calls allowguest=no ; Allow or reject guest calls -sin password- (default is yes) allowoverlap=no ; Disable overlap dialing support. (Default is yes) udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all) tcpenable=yes; Enable server for incoming TCP connections (default is no) tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces) srvlookup=yes ; Enable DNS SRV lookups on outbound calls relaxdtmf=yes dtmfmode=inband ;rfc2833compensate=yes [sip.ericsson] ;cambios allowguest hosts allowguest=no ; Allow or reject guest calls -sin password- (default is yes) type=friend calllimit=200 fromuser=ivr1 dtmfmode=inband username=administrador context=incoming-sip-ericsson host=10.146.9.70 host=ericsson host=MSSASU1.MYDOMAIN.COM.PY port=5060 disallow=all allow=alaw allow=gsm allow=ulaw qualify=yes insecure=no From: rafael_vis...@hotmail.com To: asterisk-users@lists.digium.com Date: Sun, 26 Aug 2012 19:52:43 -0400 Subject: Re: [asterisk-users] the lenght of the uri affects on dialplan? Ok... sip.conf [general] context=default ; Default context for incoming calls allowguest=no ; Allow or reject guest calls -sin password- (default is yes) allowoverlap=no ; Disable overlap dialing support. (Default is yes) udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all) tcpenable=yes; Enable server for incoming TCP connections (default is no) tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces) srvlookup=yes ; Enable DNS SRV lookups on outbound calls relaxdtmf=yes dtmfmode=inband ;rfc2833compensate=yes users.conf [general] fullname = New User userbase = 6000 hasvoicemail = yes vmsecret = 1234 hassip = yes hasiax = no hash323 = no hasmanager = no callwaiting = yes threewaycalling = yes callwaitingcallerid = yes transfer = yes canpark = yes cancallforward = yes callreturn = yes callgroup = 1 pickupgroup = 1 allowguest=no ; Allow or reject guest calls -sin password- (default is yes) [sip.ericsson] ;cambios allowguest hosts ;allowguest=no ; Allow or reject guest calls -sin password- (default is yes) type=friend calllimit=200 fromuser=ivr1 dtmfmode=inband username=administrador context=incoming-sip-ericsson host=10.146.9.70 host=ericsson host=MSSASU1.MYDOMAIN.COM.PY port=5060 disallow=all allow=alaw allow=gsm allow=ulaw qualify=yes insecure=no Date: Mon, 27 Aug 2012 03:42:51 +0500 From: fai...@vopium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] the lenght of the uri affects on dialplan? mention the complete scnario and your sip.conf. Regards, Faisal (sent from phone) Rafael Visser rafael_vis...@hotmail.com wrote: Hi Gurus.. I use asterisk for just for ivr. My issue is that when the switch changes it's host name from MSSASU1.MYDOMAIN to MSSSASU1.MYDOMAiN.COM or MSSSASU1.MYDOMAiN.COM.PY the call is rejected with No matching peer and the handle_request_invite: Sending fake auth rejection for device x. It doesn't match it's own default context. Also, it has somethig to do with the numbers of digits of the dialed number. Few digits works ok, 14 to more works wrong. Do you know what am i missing? Thanks in advance. Debug with long hostname (B is considered as an '*') --- SIP read from TCP:10.146.9.70:6240 --- INVITE sip:b56510123456789012...@sisivr03.mydomain.com.py;user=phone SIP/2.0 From: sip:971200...@mssasu1.mydomain.com.py;user=phone;tag=3016589695 To: sip:b56510123456789012...@sisivr03.mydomain.com.py;user=phone Max-Forwards: 70 Via: SIP/2.0/TCP MSSASU1.MYDOMAIN.COM.PY:5060;branch=z9hG4bK0035391821780096 Call-ID: 9cax8060616182201-bo...@mssasu1.mydomain.com.py CSeq: 7313 INVITE P-Asserted-Identity: sip:971200...@mssasu1.mydomain.com.py;user=phone Accept: application/sdp Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,PRACK,UPDATE P-Charging-Vector: icid-value=6510081000-0826-16155907;icid-generated-at=MSSASU1.MYDOMAIN.COM.PY;orig-ioi=MSSASU1.MYDOMAIN.COM.PY Supported: 100rel Content-Type: application/sdp Contact: sip:MSSASU1.MYDOMAIN.COM.PY:5060;transport=UDP Content-Length: 414 v=0 o=- 7530078 7530078 IN IP4 MSSASU1.MYDOMAIN.COM.PY s=- t=0 0 a=sendrecv m=audio 13802 RTP/AVP 8 96 18 97 c=IN IP4 10.143.1.67 b=RR:0 b=RS:0 a=rtpmap:8 PCMA/8000 a=rtpmap:96 AMR/8000 a=fmtp:96 mode-set=0,2,4,7;mode-change-period=2;mode-change-capability=2;mode-change-neighbor=1;max-red=0 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-15 a=maxptime:40
Re: [asterisk-users] SSM
You can do it with perl, using agi and smpp modules. rv 2012/5/20 CDR vene...@gmail.com: I need to send SMS from Asterisk to an SMPP server. Is there a SMPP channel or any other know way to send SMS via Asterisk? I don't care if its is paid software. Philip -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] special digits * # on sip dial string
hi guys. sorry if this is a silly question. My recharge application uses * digits if the subscriber wants to send some aditional information to speed up a process, dialing something like *777*123*5000 On my old ss7 network works great, but on my new ngn/sip i think it's not possible because somewhere the call is rejected. -On the NGN/Ericsson side engineer say that the call whas deliverd. -On the asterisk side there is no invite shown on debug. Can sip one or more * signs in a dial? What am i doing wrong. thanks in advance.. rv -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unknown Agent Status on DAHDI
Hi Guys: I am very new in Asterisk Queue, so may be i'm doing wrong somewhere. I have Asterisk 1.8.3.3 and Dahdi 2.4.1.2. I defined some agent's on Asterisk Queue, and the problem is that the agent is allways on UNKNOWN status, so Asterisk can dial to the agent even if the agent is allready busy. No matter if the agent is dynamic, realtime or static. I tried with sip channels and there where no problems, the problem is only with dahdi. Do you have any tips for this issue?. Sorry if i am the wrong list. thanks in advance. rv -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI Error
I didn't understand very well.. So you cant dial on the first 24 channels? Did you take care on the jumper of the card?. There is something related to E1 (31 channels) or T1 (24 channels). And check the system.conf either. rv 2011/5/13 deeps backup backup.de...@gmail.com I have checked destination numbers are correct as otherwise calls to those numbers are connecting fine. I opened verbose logs and digged into it more. I found out can’t dial any channels from DAHDI/24 on first E1. Before that channel calls are going through fine. I tried test calls to second E1 and can’t dial on it either. When I check channel or E1 status it is showing fine. Checked chan_dahdi and system conf files and see all channels are configured fine. Could you please help? On 13 May 2011 15:07, deeps backup backup.de...@gmail.com wrote: On 13 May 2011 14:06, Eric Wieling ewiel...@nyigc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of deeps backup Sent: Friday, May 13, 2011 9:02 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] DAHDI Error Hi, Sometimes calls on Asterisk fail to connect to DAHDI channels and giving below error: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) There are 8 E1 connected on server and only 15-20 simultaneous calls. All channels and E1 are showing in service without any alarms. Could anyone please let me know why this is happening? The message is likely coming from the telco or from the destination number. It is a common issue. I usually put something in my dialplan to retry all calls that receive an unexpected hangup cause to work around the telco seemingly randomly sending back odd hangup causes. You should not retry ALL calls, only ones with unexpected hangup causes. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I have checked destination numbers are correct as otherwise calls to those numbers are connecting fine. I opened verbose logs and digged into it more. I found out can’t dial any channels from DAHDI/24 on first E1. Before that channel calls are going through fine. I tried test calls to second E1 and can’t dial on it either. When I check channel or E1 status it is showing fine. Checked chan_dahdi and system conf files and see all channels are configured fine. Could you please help? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk 1.2 and Dial with LIMIT_WARNING_FILE
Hi fellows..
I have 2 asterisk servers in which the following line
exten = _09049.,111,SetVar(LIMIT_PLAYAUDIO_CALLER=YES)
exten = _09049.,112,SetVar(LIMIT_WARNING_FILE=beep)
exten = _09049.,113,Dial(${TYPE}${DESTINO}|30|L(3:1))
works OK on my Asterisk 1.2.9, it plays the beep 10 seconds before the
end of the call.
doesn't work on my Asterisk 1.2.13, it hungs 10 seconds before the end
of the call, just when it has to beep
both of them have the same chan_ss7 and the beep.gsm in the correct place.
Do you have any idea of what is happening?
Thanks in advance..
Rafael Visser
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[Asterisk-Users] enablling Te110p with PRI
Hi gurus... I have connected an asterisk with a te110p/pri to a GSM ericsson switch, all apears to be write. But when i try to make an outbond call from asterisk to the te110p group, the folowing error is logged: -- Executing Dial(SIP/201-5923, ZAP/1-1/0971200152|20|r) in new stack == Everyone is busy/congested at this time (1:0/0/1) Question: Is there a how to connect the Asterisk to an ericsson sw? What other test can i do against the switch?. Thanks in advance... this is the te110p configuration... asterisk1*CLI pri show span 1 Primary D-channel: 16 Status: Provisioned, Down, Active Switchtype: EuroISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 asterisk1*CLI zap show channel 1 Channel: 1CLI File Descriptor: 19 Span: 1k1*CLI Extension: LI Dialing: noLI Context: default Caller ID: Calling TON: 0 Caller ID name: Destroy: 0 InAlarm: 1 Signalling Type: PRI Signalling Radio: 0 Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: alaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps, currently OFF PRI Flags: PRI Logical Span: Implicit Hookstate (FXS only): Onhook asterisk1*CLI [EMAIL PROTECTED] asterisk]# cat /proc/zaptel/1 Span 1: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 HDB3/CCS/CRC4 1 WCT1/0/1 Clear (In use) 2 WCT1/0/2 Clear (In use) 3 WCT1/0/3 Clear (In use) 4 WCT1/0/4 Clear (In use) 5 WCT1/0/5 Clear (In use) 6 WCT1/0/6 Clear (In use) 7 WCT1/0/7 Clear (In use) 8 WCT1/0/8 Clear (In use) 9 WCT1/0/9 Clear (In use) 10 WCT1/0/10 Clear (In use) 11 WCT1/0/11 Clear (In use) 12 WCT1/0/12 Clear (In use) 13 WCT1/0/13 Clear (In use) 14 WCT1/0/14 Clear (In use) 15 WCT1/0/15 Clear (In use) 16 WCT1/0/16 HDLCFCS (In use) 17 WCT1/0/17 Clear (In use) 18 WCT1/0/18 Clear (In use) 19 WCT1/0/19 Clear (In use) 20 WCT1/0/20 Clear (In use) 21 WCT1/0/21 Clear (In use) 22 WCT1/0/22 Clear (In use) 23 WCT1/0/23 Clear (In use) 24 WCT1/0/24 Clear (In use) 25 WCT1/0/25 Clear (In use) 26 WCT1/0/26 Clear (In use) 27 WCT1/0/27 Clear (In use) 28 WCT1/0/28 Clear (In use) 29 WCT1/0/29 Clear (In use) 30 WCT1/0/30 Clear (In use) 31 WCT1/0/31 Clear (In use) AVISO LEGAL: Esta información es privada y confidencial y está dirigida únicamente a su destinatario. Si usted no es el destinatario original de este mensaje y por este medio pudo acceder a dicha información por favor elimine el mensaje. La distribución o copia de este mensaje está estrictamente prohibida. Esta comunicación es sólo para propósitos de información y no debe ser considerada como propuesta, aceptación ni como una declaración de voluntad oficial de NUCLEO S.A. La transmisión de e-mails no garantiza que el correo electrónico sea seguro o libre de error. Por consiguiente, no manifestamos que esta información sea completa o precisa. Toda información está sujeta a alterarse sin previo aviso. . This information is private and confidential and intended for the recipient only. If you are not the intended recipient of this message you are hereby notified that any review, dissemination, distribution or copying of this message is strictly prohibited. This communication is for information purposes only and shall not be regarded neither as a proposal, acceptance nor as a statement of will or official statement from NUCLEO S.A. . Email transmission cannot be guaranteed to be secure or error-free. Therefore, we do not represent that this information is complete or accurate and it should not be relied upon as such. All information is subject to change without notice. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] enablling Te110p with PRI
Sorry, at the switch side, the spam is ABL (automatic blocked), is like to be Not Aligned. Im not sure about the required parameters to configure the ericsson with isdn-pri. So, lets just wait if someone help me with the isdn config first.. Thanks. Steven Ringwald [EMAIL PROTECTED]@lists.digium.com con fecha 20/04/2006 03:47:06 p.m. Por favor, responda a [EMAIL PROTECTED]; Por favor, responda a Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Enviado por: [EMAIL PROTECTED] Destinatarios:Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com CC: Asunto: Re: [Asterisk-Users] enablling Te110p with PRI Rafael Visser wrote: Hi gurus... I have connected an asterisk with a te110p/pri to a GSM ericsson switch, all apears to be write. But when i try to make an outbond call from asterisk to the te110p group, the folowing error is logged: -- Executing Dial(SIP/201-5923, ZAP/1-1/0971200152|20|r) in new stack == Everyone is busy/congested at this time (1:0/0/1) Question: Is there a how to connect the Asterisk to an ericsson sw? What other test can i do against the switch?. Thanks in advance... What does the exact Dial line look like in your extensions.conf? Is 0971200152 the number that the other end is expecting? For instance, our Shoretel requires the country code be added, for instance 1503XXX. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Visite nuestro Sitio http://www.personal.com.py AVISO LEGAL: Esta información es privada y confidencial y está dirigida únicamente a su destinatario. Si usted no es el destinatario original de este mensaje y por este medio pudo acceder a dicha información por favor elimine el mensaje. La distribución o copia de este mensaje está estrictamente prohibida. Esta comunicación es sólo para propósitos de información y no debe ser considerada como propuesta, aceptación ni como una declaración de voluntad oficial de NUCLEO S.A. La transmisión de e-mails no garantiza que el correo electrónico sea seguro o libre de error. Por consiguiente, no manifestamos que esta información sea completa o precisa. Toda información está sujeta a alterarse sin previo aviso. . This information is private and confidential and intended for the recipient only. If you are not the intended recipient of this message you are hereby notified that any review, dissemination, distribution or copying of this message is strictly prohibited. This communication is for information purposes only and shall not be regarded neither as a proposal, acceptance nor as a statement of will or official statement from NUCLEO S.A. . Email transmission cannot be guaranteed to be secure or error-free. Therefore, we do not represent that this information is complete or accurate and it should not be relied upon as such. All information is subject to change without notice. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
