Re: [asterisk-users] Odd Local Channel and 0 billsec issue
I only got this the other day, wont go into why it's taken so long though...
[Nov 20 10:15:25] VERBOSE[3946] logger.c: -- Executing
Answer(SIP/IP-ADDRESS-c404b3c0, )
[Nov 20 10:15:25] VERBOSE[3946] logger.c: -- Executing
Wait(SIP/IP-ADDRESS-c404b3c0, 2)
[Nov 20 10:15:27] VERBOSE[3946] logger.c: -- Executing
Set(SIP/IP-ADDRESS-c404b3c0, CALLERID(num)=NUMBER)
[Nov 20 10:15:27] VERBOSE[3946] logger.c: -- Executing
Dial(SIP/IP-ADDRESS-c404b3c0, SIP/AOGW100SIP/AOGW101|20)
[Nov 20 10:15:27] VERBOSE[3946] logger.c: -- Called AOGW100
[Nov 20 10:15:27] VERBOSE[3946] logger.c: -- Called AOGW101
[Nov 20 10:15:27] VERBOSE[29370] logger.c: -- Got SIP response 302
Moved Temporarily back from IPADDRESS
[Nov 20 10:15:27] VERBOSE[3946] logger.c: -- Now forwarding
SIP/IP-ADDRESS-c404b3c0 to 'Local/dialled-num...@aogw-local' (thanks to
SIP/AOGW100-0095a6b0)
[Nov 20 10:15:27] VERBOSE[3949] logger.c: -- Executing
Macro(Local/dialled-num...@aogw-local-ecb8,2,
extcall|DIALLED-NUMBER|CLID)
[Nov 20 10:15:27] VERBOSE[3949] logger.c: -- Executing
[...@macro-extcall:1] Set(Local/dialled-num...@aogw-local-ecb8,2,
CALLERID(all)=CLID) in new stack
[Nov 20 10:15:27] VERBOSE[3949] logger.c: -- Executing
[...@macro-extcall:2] Dial(Local/dialled-num...@aogw-local-ecb8,2,
SIP/dialled-num...@magrathea) in new stack
[Nov 20 10:15:27] VERBOSE[3949] logger.c: -- Called
dialled-num...@magrathea
[Nov 20 10:15:27] VERBOSE[3946] logger.c: -- SIP/AOGW101-00aa80d0 is
ringing
I've obscured some of the sensitive data.
I would have thought there is a forward set up from one of the handsets
but why would it use a Local channel, I've tested this before and it
used a SIP channel in all of my tests.
Ish
Warren Selby wrote:
CLI output of calls that go through the local channel instead of the
defined channel would be useful to help diagnose what's going on here.
Thanks,
--Warren Selby
On Mon, Nov 16, 2009 at 4:01 AM, Ishfaq Malik i...@pack-net.co.uk
mailto:i...@pack-net.co.uk wrote:
Hi
I've been noticing an odd issue with our servers (1.4.17) where a
large
number of one particular customer's (we operate a hosted VoIP
platform)
calls go through a Local channel rather than the SIP channel and
whenever this happens our asterisk CDR is recording a billsec
value of 0.
Our outgoing calls to POTS are sent through a separate carrier and we
get a daily CDR off them in which these same calls have a non 0
duration
so we are obviously making a loss on these calls.
All out customers outgoing calls go through the same macro which is as
follows
[macro-extcall]
;Macro created by Ish to handle external calls
exten = s,1,Set(CALLERID(all)=${ARG2})
exten = s,2,Dial(SIP/44${ar...@carrier)
exten = s,3,Hangup
exten = s,102,Playtones(busy)
exten = s,103,Congestion
I've seen the same issue very occasionally with other customers
but with
one particular customer a large proportion, but not all the calls show
this issue.
Has anyone had any experience of similar issues?
Thanks in advance
Ish
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
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Thanks,
--Warren Selby
http://www.selbytech.com
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Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
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[asterisk-users] Is Answer really needed
Hi All my incoming dial plans start of with an Answer which I now know starts the billing time. Some of the dialplans then get forwarded out to POTS via a carrier and so the actual amount of time that should be billed is being distorted. I've done a few tests this morning and found that if I don't start with an answer then the billsec of my forwarded call is actually the length of time that the call was answered in reality and not the length of the call plus connection and answering time. My question is, does Answer have any major function that I am overlooking before I remove it from all my dial plans? Because it doesn't seem to to me. Thanks Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Answer really needed
Recall that in regards to SIP implementation, Asterisk is a back-to-back user agent (B2BUA). This means that one logical call leg comes in, and another logical call leg is generated out, and the two are cross-connected. If SIP is not the signaling technology used on one or both channels, the effect is analogical where applicable. However, I will use SIP to illustrate the point; you can extrapolate from there similar effects on other channel types. The function that Answer() has on a signaling level is to effect an pickup on the incoming call leg. In SIP, this is a 200 OK message. If you then proceed to Dial() out on another channel, any ringback generated out the first channel will be in-band; that is to say, it will be inside the acoustic bearer. A far-end pickup (200 OK) is necessary to exchange audio bidirectionally. Some dial plan functions - mostly those that conceivably entail a two-way communication path - imply Answer() and will execute it for you if you have not already done so. Others do not. For example, it is possible to generate in-band ringback via early media, e.g. by sending a 183 Session in Progress message with an SDP payload to the sender. So, for example, if you were to do this: exten = s,1,MusicOnHold without doing an Answer() first, the MOH would be played via early media without pickup. By the same token, if you Dial() out before Answer()ing, the ringback generated will also be via early media (or, if applicable, out-of-band, depending on other settings): exten = s,1,Dial(SIP/otherpl...@other_peer) This will not result in a 200 OK received on the far end of the incoming channel until there is a 200 OK received on the near end of the outgoing channel. That is the function that Answer() serves. The option to remove it is contingent upon refraining from use of dial plan applications that implicitly invoke it. -- Alex Ishfaq Malik wrote: Hi All my incoming dial plans start of with an Answer which I now know starts the billing time. Some of the dialplans then get forwarded out to POTS via a carrier and so the actual amount of time that should be billed is being distorted. I've done a few tests this morning and found that if I don't start with an answer then the billsec of my forwarded call is actually the length of time that the call was answered in reality and not the length of the call plus connection and answering time. My question is, does Answer have any major function that I am overlooking before I remove it from all my dial plans? Because it doesn't seem to to me. Thanks Ish -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connect two Asterisk Server in IAX ?
Hi,
maybe this link can be useful:
http://www.voip-info.org/wiki/view/IAX+encryption
In particular, in your configuration I can't see the authentication method,
which must be md5, and a username to authenticate with, in either server.
But have a further look at the article, maybe you'll be able to sort out the
issue from that :)
HTH
//Al.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Phibee Network
Operation Center
Sent: sabato 21 novembre 2009 8.16
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Connect two Asterisk Server in IAX ?
Hi
My first post get no answer :=, i post new with new elements.
I have two Asterisk server, running on Asterisk 1.6:
SRV1 = 192.168.0.5 on Asterisk 1.6.1.4
SRV2 = 192.168.0.20 on Asterisk 1.6.1.8
I want create a link for exchange call.
on Srv1:
iax.conf:
[general]
bindport=4569
bindaddr=0.0.0.0
language=fr
bandwidth=low
jitterbuffer=no
forcejitterbuffer=no
autokill=yes
calltokenoptional=192.168.0.20
[Srv2]
type=peer
host=192.168.0.20
qualify=yes
trunk=no
encryption=aes128
disallow=all
allow=alaw
allow=g729
context=Incoming
peercontext=Incoming
extension.conf:
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
[globals]
CONSOLE=Console/dsp ; Console interface for demo
[Incoming]
exten = _X.,1,Playback(demo-thanks)
exten = _X.,2,Hangup
[Out]
exten = _201X.,1,Dial(IAX2/Srv2/${EXTEN:3},90,r)
exten = _201X.,2,Congestion
==
Srv1*CLI iax2 show peers
Name/UsernameHost Mask Port Status
Srv2 192.168.0.20 (S) 255.255.255.255 4569 (E) OK (39 ms)
1 iax2 peers [1 online, 0 offline, 0 unmonitored]
On Srv2
iax.conf
[general]
bindport=4569
bindaddr=0.0.0.0
language=fr
bandwidth=low
jitterbuffer=no
forcejitterbuffer=no
autokill=yes
calltokenoptional=192.168.0.5
bandwidth=low
[Srv1]
type=peer
host=192.168.0.5
qualify=yes
trunk=no
encryption=aes128
disallow=all
allow=alaw
allow=g729
context=Incoming
peercontect=Incoming
extensions.conf:
[Incoming]
exten = _X.,1,Playback(demo-thanks)
exten = _X.,2,Hangup
[Out]
exten = _202X.,1,Dial(IAX2/Srv1/${EXTEN:3},90,r)
exten = _202X.,2,Congestion
===
trader-voip*CLI iax2 show peers
Name/UsernameHost Mask Port Status
Srv1 192.168.0.5 (S) 255.255.255.255 4569 (E) OK (28 ms)
1 iax2 peers [1 online, 0 offline, 0 unmonitored]
===
All SIP Poste are connected and have in context in: Out
Now, when i call from a post connected on Srv1, i have this error on Srv1:
[Nov 21 08:09:44] WARNING[6407]: chan_iax2.c:9018 socket_process: Call
rejected by 192.168.0.20: No authority found
and on Srv2:
[Nov 21 08:09:44] NOTICE[9089]: chan_iax2.c:9785 socket_process:
Rejected connect attempt from 192.168.0.5, who was trying to reach
'1...@incoming'
125 are the number called (201125)
Dialplan on Srv2
Srv2*CLI dialplan show Incoming
[ Context 'Incoming' created by 'pbx_config' ]
'_X.' = 1. Playback(demo-thanks)
[pbx_config]
2. Hangup()
[pbx_config]
-= 1 extension (2 priorities) in 1 context. =-
Anyone can help me for know where is my error ?
thanks
Jerome
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[asterisk-users] Get the extension dailed
Hello When a user makes a call to an Asterisk system, He dials a number . We need to know that dialed number. We can get the dialed number by using CALLERID(dnid) and we can get the CLI information using CALLERID(num). I am facing problem while getting the number dialed. if the user is using SIP phone then we can get the number dialed. but if it using PSTN then we are unable to get the number dialed using CALLERID(dnid). Do any other way exists to find out? Or this is some thing wrong with the PSTN. -- Kind Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Get the extension dailed
ABBAS SHAKEEL wrote:
I am facing problem while getting the number dialed. if the user is
using SIP phone then we can get the number dialed. but if it using PSTN
then we are unable to get the number dialed using CALLERID(dnid). Do any
other way exists to find out? Or this is some thing wrong with the PSTN.
DNID breakage is a long-standing Asterisk problem.
If this is taking place in the context of the dial plan, why not just
use ${EXTEN}?
--
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
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Re: [asterisk-users] Get the extension dailed
Thanks Alex,
suppose this is the context
[abc]
exten = s,1,Answer();
exten = s,n,Noop(${EXTEN});
exten = s,n,Noop(${CALLERID(dnid)});
I get the following out put
Answer(DAHDI/2-1, )
NoOp(DAHDI/2-1, s) in new stack
NoOp(DAHDI/2-1, )
But i need the number that the user is dialing not the s;
On Mon, Nov 23, 2009 at 3:24 PM, Alex Balashov abalas...@evaristesys.comwrote:
ABBAS SHAKEEL wrote:
I am facing problem while getting the number dialed. if the user is
using SIP phone then we can get the number dialed. but if it using PSTN
then we are unable to get the number dialed using CALLERID(dnid). Do any
other way exists to find out? Or this is some thing wrong with the PSTN.
DNID breakage is a long-standing Asterisk problem.
If this is taking place in the context of the dial plan, why not just
use ${EXTEN}?
--
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
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Kind Regards
Shakeel Abbas
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Re: [asterisk-users] Get the extension dailed
I am curious what happens if you do the following instead:
[abc]
exten = _.,1,Answer
exten = _.,n,NoOp(${EXTEN})
ABBAS SHAKEEL wrote:
Thanks Alex,
suppose this is the context
[abc]
exten = s,1,Answer();
exten = s,n,Noop(${EXTEN});
exten = s,n,Noop(${CALLERID(dnid)});
I get the following out put
Answer(DAHDI/2-1, )
NoOp(DAHDI/2-1, s) in new stack
NoOp(DAHDI/2-1, )
But i need the number that the user is dialing not the s;
On Mon, Nov 23, 2009 at 3:24 PM, Alex Balashov
abalas...@evaristesys.com mailto:abalas...@evaristesys.com wrote:
ABBAS SHAKEEL wrote:
I am facing problem while getting the number dialed. if the user is
using SIP phone then we can get the number dialed. but if it
using PSTN
then we are unable to get the number dialed using CALLERID(dnid).
Do any
other way exists to find out? Or this is some thing wrong with
the PSTN.
DNID breakage is a long-standing Asterisk problem.
If this is taking place in the context of the dial plan, why not just
use ${EXTEN}?
--
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
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Kind Regards
Shakeel Abbas
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Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
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Re: [asterisk-users] Get the extension dailed
Thanks Alex
Thats just a test code
exten = _.,1,Answer
exten = _.,n,NoOp(${EXTEN})
This also have the same output as that of previous.
On Mon, Nov 23, 2009 at 3:46 PM, Alex Balashov abalas...@evaristesys.comwrote:
I am curious what happens if you do the following instead:
[abc]
exten = _.,1,Answer
exten = _.,n,NoOp(${EXTEN})
ABBAS SHAKEEL wrote:
Thanks Alex,
suppose this is the context
[abc]
exten = s,1,Answer();
exten = s,n,Noop(${EXTEN});
exten = s,n,Noop(${CALLERID(dnid)});
I get the following out put
Answer(DAHDI/2-1, )
NoOp(DAHDI/2-1, s) in new stack
NoOp(DAHDI/2-1, )
But i need the number that the user is dialing not the s;
On Mon, Nov 23, 2009 at 3:24 PM, Alex Balashov
abalas...@evaristesys.com mailto:abalas...@evaristesys.com wrote:
ABBAS SHAKEEL wrote:
I am facing problem while getting the number dialed. if the user
is
using SIP phone then we can get the number dialed. but if it
using PSTN
then we are unable to get the number dialed using CALLERID(dnid).
Do any
other way exists to find out? Or this is some thing wrong with
the PSTN.
DNID breakage is a long-standing Asterisk problem.
If this is taking place in the context of the dial plan, why not just
use ${EXTEN}?
--
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
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Kind Regards
Shakeel Abbas
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Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
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Kind Regards
Shakeel Abbas
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[asterisk-users] Interconnect Asterisk with another PBX
Hi, I want to interconnect a Alcatel OmniPCX PBX with SIP support with an Asterisk PBX. My intention is Alcatel PBX manage all external calls and analog extensions and Asterisk manage all the SIP users (because I have to pay for every SIP license in Alcatel PBX and I can’t edit configuration or password in that PBX) What’s the best way to interconnect the 2 PBX? With SIP, with a FXO interface or FXS? How can I do that? Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Interconnect Asterisk with another PBX
PRI is likely the simplest and most reliable. Xavier Mesquida wrote: Hi, I want to interconnect a Alcatel OmniPCX PBX with SIP support with an Asterisk PBX. My intention is Alcatel PBX manage all external calls and analog extensions and Asterisk manage all the SIP users (because I have to pay for every SIP license in Alcatel PBX and I can’t edit configuration or password in that PBX) What’s the best way to interconnect the 2 PBX? With SIP, with a FXO interface or FXS? How can I do that? Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Connect Two Asterisk's using isdn Cards
Hi, all For some work i'am trying to connect to Asterisk's PBX using isdn cards. But I don't know anything about it. So i'll be pleased to get some information about it! Thanks, Best rgrds!! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connect Two Asterisk's using isdn Cards
On Mon, Nov 23, 2009 at 06:49:37AM -0600, mos...@infolog.mr wrote: Hi, all For some work i'am trying to connect to Asterisk's PBX using isdn cards. Which cards exactly? But I don't know anything about it. So i'll be pleased to get some information about it. http://voip-info.org/ http://asteriskdocs.org/ http://www.asterisk.org/docs -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connect Two Asterisk's using isdn Cards
On 23 Nov 2009, at 12:49, mos...@infolog.mr wrote: For some work i'am trying to connect to Asterisk's PBX using isdn cards. Are you trying to connect 'to' Asterisk PBXs or 'two' Asterisk PBXs? If its 'two' then why use ISDN? Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Please some enlightment on ENUM !!
Hello all you Gurus out there! Please could you explain something to me: Currently I try to get ENUMLOOKUP() working. Naturally I do all the testing with my own number. I registered my number at e164.org I paid for registration of my number at a registration agent for e164.arpa (I know, I don't need both. I just did the .arpa registration first and later discoverd the free .org service) Assume my number was +4311234567 dig 7.6.5.4.3.2.1.1.3.4.e164.org and dig 7.6.5.4.3.2.1.1.3.4.e164.arpa both return NAPTR records. Now for the less clearer points: Your'e supposed to register your number without any extension. If I have some extensions here, how can the calling party get the correct sip uri to the requested extension? Do I have to run my own DNS server in that case? If for example if someone wants to call extension 10, is the ENUMLOOKUP(431123456710) request forwarded to my local DNS server by the e164.arpa server? Or how does that work? Many thanks, Norbert ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.6.1.10 Music On Hold
Hello. I just upgraded from 1.6.0.9 to 1.6.1.10 and it seems that the Music On Hold functionality has changed (or is bugged?). I have Aastra 6757i and Aastra 6731i phones, and now when i press the MusicOnHold button / change lines on the phone, MOH no longer starts. It did this in v 1.6.0.9. The invites received are exactly the same, only 1.6.1.10 doesn't ever start MOH. Is there some configuration change I need to implement for this to work properly? Was there a conscious change in Asterisk's behavior? Best regards, Örn ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please some enlightment on ENUM !!
- Original Message - From: Norbert Zawodsky To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, November 23, 2009 3:15 PM Subject: [asterisk-users] Please some enlightment on ENUM !! Hello all you Gurus out there! Please could you explain something to me: Currently I try to get ENUMLOOKUP() working. Naturally I do all the testing with my own number. I registered my number at e164.org I paid for registration of my number at a registration agent for e164.arpa (I know, I don't need both. I just did the .arpa registration first and later discoverd the free .org service) Assume my number was +4311234567 dig 7.6.5.4.3.2.1.1.3.4.e164.org and dig 7.6.5.4.3.2.1.1.3.4.e164.arpa both return NAPTR records. Now for the less clearer points: Your'e supposed to register your number without any extension. If I have some extensions here, how can the calling party get the correct sip uri to the requested extension? Do I have to run my own DNS server in that case? If for example if someone wants to call extension 10, is the ENUMLOOKUP(431123456710) request forwarded to my local DNS server by the e164.arpa server? Or how does that work? If everybody supported enum, it might be usefull to register extension 10 in enum, otherwise: Your extension 10 must have its own phonenumber, to be able to dial it directly. Just as with ordinary pabx. Eg: 123 555 is the reception 123 555 0010 is extension 10 Just some ideas: Is there free (as in not connected to a voisp) numbers, which can be registered in enum? Then you could use those numbers for extensions. But they would only be callable by enum. If the calling of extensions is only to be used by knowledgeable friends you could have them add your own enum-domain to their setup. Leif ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Questions about Voicemail
I am sorry if this is not the correct place to post a question. I am wondering if there is way in asterisk 1.2 to: 1. Send a voicemail (from the phone) to multiple recipients. 2. Require (as an admin) for users 1st logging into their voicemail to change their greeting and/or password. Thanks --Dovey ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meetme 'o' - what actually it does..??
On Mon, Nov 23, 2009 at 2:17 AM, Chandrakant Solanki solanki.chandrak...@gmail.com wrote: Hi Can someone explain me what is the purpose for MeetMe Option 'o'.. If I defined 'o' with MeetMe option or If not defined with MeetMe option... What is the difference between these two if defined or not defined MeetMe 'o' option... Well, there's theory, and then there's my experience... Theory says that the larger the conference, the more that people introduce small noises, like ambient hums, and these eventually become a lot of the mixing load on a conference, degrading the experience for all. By trying to tell the difference between speech and ambient noise, a conference can do a better job of making a conference sound good for all users. This is what is meant by talker optimization. In my usage, the optimization was too optimal, and was clipping the beginnings and endings of sentences and phrases, and I upgraded to get a release where optimization was off by default and optional. In my usage, we're mostly using small conferences. It's entirely possible that with large conferences the optimization is very useful. I've never used it that way, so I can't say myself. There was a large demand to make talker optimization optional again, after 1.6.0 initially launched with optimization always-on, thus I don't think I'm the only one who tried optimization and thought it made things worse than without the optimization. It's also possibly that optimization is better now than it used to be. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Interconnect Asterisk with another PBX
Either use SIP or PRIs to do the integration. FXO and FXS interfaces are a single port, where as a PRI will provide you with 23 channels. Use QSIG signaling over the PRI so Caller ID names will show between the systems. I just integrated a Toshiba CIX with Asterisk due to the cost for SIP licensing and the reliability of the Toshiba VOIP Phones. They were having hardware failures every few months. I went with Sangoma PRI cards using QSIG. Everything has been working great and I have rolled out 12 Snom 370 phones to work with the 150 Toshiba Digital phones. To the end users the experience is seamless as they can 4 digit dial any extension and the call will be routed to the correct system. This does take a bit of duplicate setup on the two systems, but was worth the hassle for the end result. Ryan On Mon, Nov 23, 2009 at 6:17 AM, Alex Balashov abalas...@evaristesys.com wrote: PRI is likely the simplest and most reliable. Xavier Mesquida wrote: Hi, I want to interconnect a Alcatel OmniPCX PBX with SIP support with an Asterisk PBX. My intention is Alcatel PBX manage all external calls and analog extensions and Asterisk manage all the SIP users (because I have to pay for every SIP license in Alcatel PBX and I can’t edit configuration or password in that PBX) What’s the best way to interconnect the 2 PBX? With SIP, with a FXO interface or FXS? How can I do that? Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] best channel driver for 1.4.x and beronet/junghanns 4BRI?
Hi, What is the best channel driver to use asterisk 1.4.x with a 4BRI isdn card from Beronet or Junghanns (same hardware, different pcid)? Are these cards now supported by plain (non-patched) dahdi/zaptel modules? Thanks, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4 and kernel panic and IRQ interrupts
Hi, I'm having trouble with one machine that kernel panics with Asterisk 1.4. The motherboard is an Asus P5W Deluxe. I reported the kernel panic here: http://lists.digium.com/pipermail/asterisk-users/2009-November/241006.html I'm now trying to understand if the problem can be an IRQ issue or not. I disabled APIC in the BIOS because I thought that maybe it could be buggy (not sure though). My interrupts are now as follows: # more /proc/interrupts CPU0 CPU1 CPU2 CPU3 0: 93 0 0 0XT-PIC-XTtimer 1: 1531 0 0 0XT-PIC-XTi8042 2: 0 0 0 0XT-PIC-XTcascade 3: 0 0 0 0XT-PIC-XTuhci_hcd:us b3 5:4012524 0 0 0XT-PIC-XTehci_hcd:us b1, uhci_hcd:usb2 6: 3 0 0 0XT-PIC-XTfloppy 7:4341422 0 0 0XT-PIC-XTahci, HFC-multi 8: 2 0 0 0XT-PIC-XTrtc 9: 1 0 0 0XT-PIC-XTacpi 10: 10306916 0 0 0XT-PIC-XTeth1, eth2 11: 30845499 0 0 0XT-PIC-XTeth0, wcte12xp0 12: 3137 0 0 0XT-PIC-XTi8042 14:213 0 0 0XT-PIC-XTide0 NMI: 0 0 0 0 LOC:3049870304985930498553049853 ERR: 0 MIS: 0 This doesn't look good for 3 reasons (I think): 1. only one core out of a quad-core CPU handles the interrupts 2. the telephony cards share IRQs with other devices (HFC-multi and wcte12xp0) 3. wcte12xp0 and eth0 are sharing the same IRQ and eth0 is particularly active on this system Note that on another system (Asus P5B motherboard with APIC enabled) I have a very stable Asterisk 1.2 and the IRQs are as follows: # cat /proc/interrupts CPU0 CPU1 CPU2 CPU3 0:104 0 0 0 IO-APIC-edge timer 1: 1558 0 0 0 IO-APIC-edge i8042 6: 3 0 0 0 IO-APIC-edge floppy 8: 2 0 0 0 IO-APIC-edge rtc 9: 1 0 0 0 IO-APIC-fasteoi acpi 16: 50387 0 0 0 IO-APIC-fasteoi ahci 17:4710977 11071200 164252433430308 IO-APIC-fasteoi ide0, eth0 18: 64081335 65109221 31317172 33363907 IO-APIC-fasteoi ahci, eth1 20: 114824294 87784625 79980388 99000512 IO-APIC-fasteoi wcte12xp0 21: 645793 0 0 0 IO-APIC-fasteoi eth2 22:94996127398138 108897606725944 IO-APIC-fasteoi HFC-multi NMI: 0 0 0 0 LOC: 37865864 37865853 37857176 37857173 ERR: 0 MIS: 0 Each telephony card is on its own IRQ. Can IRQ sharing actually cause a kernel panic? or does it usually only cause voice distortion, ticks, etc.? What do you suggest I should try? Should I enable APIC again and try to get each card on a different IRQ? Is anyone using an Asus P5W Deluxe? If so, could you please share your /proc/interrupts and BIOS settings? Thanks, Vieri ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ADSI...
Can anyone tell me the proper ADSI syntax for prompting a user to enter their password or an extension? -- Shay Smith Gmail Evangelist David Douglas School District 503-261-8235 http://www.ddouglas.k12.or.us Sent from Portland, OR, United States ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Interconnect Asterisk with another PBX
I am doing what u wanna atm but instead of an Alcatlet with SIP support i have to struggle with an Avaya CM without SIP but with H.323. So far putting a trunk over Ethernet with SIP is the way I gonna go. I havent run in to any show-stopper so far with my CM H.323 - Asterisk integration. On Mon, 23 Nov 2009 11:17:22 -0500, Ryan Wagoner wrote: Either use SIP or PRIs to do the integration. FXO and FXS interfaces are a single port, where as a PRI will provide you with 23 channels. Use QSIG signaling over the PRI so Caller ID names will show between the systems. I just integrated a Toshiba CIX with Asterisk due to the cost for SIP licensing and the reliability of the Toshiba VOIP Phones. They were having hardware failures every few months. I went with Sangoma PRI cards using QSIG. Everything has been working great and I have rolled out 12 Snom 370 phones to work with the 150 Toshiba Digital phones. To the end users the experience is seamless as they can 4 digit dial any extension and the call will be routed to the correct system. This does take a bit of duplicate setup on the two systems, but was worth the hassle for the end result. Ryan On Mon, Nov 23, 2009 at 6:17 AM, Alex Balashov wrote: PRI is likely the simplest and most reliable. Xavier Mesquida wrote: Hi, I want to interconnect a Alcatel OmniPCX PBX with SIP support with an Asterisk PBX. My intention is Alcatel PBX manage all external calls and analog extensions and Asterisk manage all the SIP users (because I have to pay for every SIP license in Alcatel PBX and I can't edit configuration or password in that PBX) What's the best way to interconnect the 2 PBX? With SIP, with a FXO interface or FXS? How can I do that? Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] GotoIfTime problem - possible bug
Hi, I'm currently doing some testing against asterisk 1.6.1.7-rc2 (keep meaning to upgrade) and am having a problem with the GotoIfTime dial plan function. The asterisk book says that day of week field can include the ampersand () to combine multiple days / day ranges but this gives me an error. For example monwed gives the error (in the asterisk console): [Nov 23 18:04:27] WARNING[11387]: pbx.c:6249 get_range: Invalid day of week 'monwed', assuming none Does anyone else have experience of this problem? Are there any patches / newer versions to get around this? Thanks in advance. Regards, Dr. Nic Colledge ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GotoIfTime problem - possible bug
On Monday 23 November 2009 12:11:02 pm Nic Colledge wrote: I'm currently doing some testing against asterisk 1.6.1.7-rc2 (keep meaning to upgrade) and am having a problem with the GotoIfTime dial plan function. The asterisk book says that day of week field can include the ampersand () to combine multiple days / day ranges but this gives me an error. For example monwed gives the error (in the asterisk console): [Nov 23 18:04:27] WARNING[11387]: pbx.c:6249 get_range: Invalid day of week 'monwed', assuming none Does anyone else have experience of this problem? Are there any patches / newer versions to get around this? That was an error on my part, when I helped review the book prior to publication. (I was incidentally thinking of the arguments to the CUT() function.) However, given that it was a good idea, it has been implemented in the forthcoming 1.6.2 release, currently in release candidate status. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GotoIfTime problem - possible bug
On Mon, Nov 23, 2009 at 1:11 PM, Nic Colledge n...@njcolledge.net wrote: I’m currently doing some testing against asterisk 1.6.1.7-rc2 (keep meaning to upgrade) and am having a problem with the GotoIfTime dial plan function. The asterisk book says that day of week field can include the ampersand () to combine multiple days / day ranges but this gives me an error. For example monwed gives the error (in the asterisk console): [Nov 23 18:04:27] WARNING[11387]: pbx.c:6249 get_range: Invalid day of week 'monwed', assuming none I use mon-fri all the time, but I've never tried to grab non-contiguous days before. Have you tried '|', as in the shift of backslash, or some people say pipe? I'm not sure I've ever actually seen documentation that says multiple non-contiguous days is valid syntax. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Questions about Voicemail
On Mon, 2009-11-23 at 10:37 -0500, Dovey Forman wrote: I am sorry if this is not the correct place to post a question. I am wondering if there is way in asterisk 1.2 to: 1. Send a voicemail (from the phone) to multiple recipients. Yes I believe so. 1. The voicemail app allows delivery to multiple destinations at once: - example : exten = 100,1,VoiceMail(u101102103) 2. Create an e-mail alias/list and deliver the voicemail via e-mail to that alias. 2. Require (as an admin) for users 1st logging into their voicemail to change their greeting and/or password. There is a user option forcegreetings: forcegreetings = [yes|no] Sets whether the user will be forced to record a new greeting when logging in to the system for the first time. Default: no Example: forcegreetings = no Not sure about the forced change PIN, but it should be easy enough to write a little command wrapper around it and prompt for PIN via the dialplan. Rob ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GotoIfTime problem - possible bug
Thanks very much, I'll fire up 1.6.2 and see how I go. Nic. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: 23 November 2009 18:20 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] GotoIfTime problem - possible bug On Monday 23 November 2009 12:11:02 pm Nic Colledge wrote: I'm currently doing some testing against asterisk 1.6.1.7-rc2 (keep meaning to upgrade) and am having a problem with the GotoIfTime dial plan function. The asterisk book says that day of week field can include the ampersand () to combine multiple days / day ranges but this gives me an error. For example monwed gives the error (in the asterisk console): [Nov 23 18:04:27] WARNING[11387]: pbx.c:6249 get_range: Invalid day of week 'monwed', assuming none Does anyone else have experience of this problem? Are there any patches / newer versions to get around this? That was an error on my part, when I helped review the book prior to publication. (I was incidentally thinking of the arguments to the CUT() function.) However, given that it was a good idea, it has been implemented in the forthcoming 1.6.2 release, currently in release candidate status. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Questions about Voicemail
Regarding the email to multiple receipients, it is available on an ad-hoc basis from the phone? IE; call into the voicemail system, enter x digit to send a voicemail to multiple users, record the message, then enter the destination mailboxes, separated by # sign... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Lister Sent: Monday, November 23, 2009 1:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Questions about Voicemail On Mon, 2009-11-23 at 10:37 -0500, Dovey Forman wrote: I am sorry if this is not the correct place to post a question. I am wondering if there is way in asterisk 1.2 to: 1. Send a voicemail (from the phone) to multiple recipients. Yes I believe so. 1. The voicemail app allows delivery to multiple destinations at once: - example : exten = 100,1,VoiceMail(u101102103) 2. Create an e-mail alias/list and deliver the voicemail via e-mail to that alias. 2. Require (as an admin) for users 1st logging into their voicemail to change their greeting and/or password. There is a user option forcegreetings: forcegreetings = [yes|no] Sets whether the user will be forced to record a new greeting when logging in to the system for the first time. Default: no Example: forcegreetings = no Not sure about the forced change PIN, but it should be easy enough to write a little command wrapper around it and prompt for PIN via the dialplan. Rob ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM400P alarm state
I'm having real problems with my connection to BT, it is a home line, but after a while it sets an alarm and only a restart of asterisk resets it could some one look at the below configs and suggest any changes to make this more reliable Thanks for your help Robb asterisk version 1.6.1.10 dahdi version SVN-trunk-r7445 dahdi_scan active=yes alarms=OK description=Wildcard TDM400P REV I Board 5 name=WCTDM/4 manufacturer=Digium devicetype=Wildcard TDM400P REV I location=PCI Bus 01 Slot 08 basechan=1 totchans=4 irq=18 type=analog port=1,FXS port=2,FXS port=3,FXS port=4,FXO system.conf # Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1 (MASTER) fxoks=1 fxoks=2 fxoks=3 fxsks=4 echocanceller=mg2,1-4 # Global data loadzone= uk defaultzone = uk # alaw=1,2,3,4 chan_dahdi.conf - channel 4 (fxs) language=en context=incomming signalling=fxs_ks ;rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=yes ;ukcallerid=yes cidsignalling=v23 ; Added for UK CLI detection cidstart=polarity ; Added for UK CLI detection restrictcid=no callerid=asreceived ;sendcalleridafter=0 polarityonanswerdelay=10 ;answeronpolarityswitch=yes ;hanguponpolarityswitch=yes resetpolarityonring=no busydetect=no callprogress=no ;progzone=uk hidecallerid=no callwaiting=no relaxdtmf=yes usecallingpres=yes callwaitingcallerid=no threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes ;echotraining=800 rxgain=3.0 txgain=3.0 group=1 immediate=yes channel = 4 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please some enlightment on ENUM !!
Leif Neland schrieb: - Original Message - *From:* Norbert Zawodsky mailto:norb...@zawodsky.at *To:* Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com *Sent:* Monday, November 23, 2009 3:15 PM *Subject:* [asterisk-users] Please some enlightment on ENUM !! Hello all you Gurus out there! Please could you explain something to me: Currently I try to get ENUMLOOKUP() working. Naturally I do all the testing with my own number. I registered my number at e164.org I paid for registration of my number at a registration agent for e164.arpa (I know, I don't need both. I just did the .arpa registration first and later discoverd the free .org service) Assume my number was +4311234567 dig 7.6.5.4.3.2.1.1.3.4.e164.org and dig 7.6.5.4.3.2.1.1.3.4.e164.arpa both return NAPTR records. Now for the less clearer points: Your'e supposed to register your number without any extension. If I have some extensions here, how can the calling party get the correct sip uri to the requested extension? Do I have to run my own DNS server in that case? If for example if someone wants to call extension 10, is the ENUMLOOKUP(431123456710) request forwarded to my local DNS server by the e164.arpa server? Or how does that work? If everybody supported enum, it might be usefull to register extension 10 in enum, otherwise: Your extension 10 must have its own phonenumber, to be able to dial it directly. Just as with ordinary pabx. Eg: 123 555 is the reception 123 555 0010 is extension 10 Just some ideas: Is there free (as in not connected to a voisp) numbers, which can be registered in enum? Then you could use those numbers for extensions. But they would only be callable by enum. If the calling of extensions is only to be used by knowledgeable friends you could have them add your own enum-domain to their setup. Leif Hi Leif! No, I cannot believe that this was the right way. It would mean that I would have to register ( pay !!) for every single extension. BTW the How-To, the registration agent I'm using provides on his website, states, that if you're operating a PBX, you should only register your main number (=without any extensions). I *assume* that if I do an ENUMLOOKUP() of a number which includes some extension at the end, the DNS request is somehow delegated to that sub-server which is authorative over this sub-domain. This leads me to the next *assumption* that the right way would be to run an own DNS server which returns the sip-uri's for my extensions. Can someone confirm this? Norbert ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP over TCP/TLS for 1.4 branch
Any word on when (or if) SIP over TCP for 1.4 branch is making an appearance? Looking to possibly do an OCS integration, but would prefer to not upgrade to 1.6 or throw OpenSer/Kamailio in the mix. Bryan Ekelund WHI Solutions, Inc. bekel...@whisolutions.com STATEMENT OF CONFIDENTIALITY: The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify WHI Solutions immediately at g...@whisolutions.com, and destroy all copies of this message and any attachments. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please some enlightment on ENUM !!
Norbert Zawodsky wrote: Leif Neland schrieb: - Original Message - *From:* Norbert Zawodsky mailto:norb...@zawodsky.at *To:* Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com *Sent:* Monday, November 23, 2009 3:15 PM *Subject:* [asterisk-users] Please some enlightment on ENUM !! Hello all you Gurus out there! Please could you explain something to me: Currently I try to get ENUMLOOKUP() working. Naturally I do all the testing with my own number. I registered my number at e164.org I paid for registration of my number at a registration agent for e164.arpa (I know, I don't need both. I just did the .arpa registration first and later discoverd the free .org service) Assume my number was +4311234567 dig 7.6.5.4.3.2.1.1.3.4.e164.org and dig 7.6.5.4.3.2.1.1.3.4.e164.arpa both return NAPTR records. Now for the less clearer points: Your'e supposed to register your number without any extension. If I have some extensions here, how can the calling party get the correct sip uri to the requested extension? Do I have to run my own DNS server in that case? If for example if someone wants to call extension 10, is the ENUMLOOKUP(431123456710) request forwarded to my local DNS server by the e164.arpa server? Or how does that work? If everybody supported enum, it might be usefull to register extension 10 in enum, otherwise: Your extension 10 must have its own phonenumber, to be able to dial it directly. Just as with ordinary pabx. Eg: 123 555 is the reception 123 555 0010 is extension 10 Just some ideas: Is there free (as in not connected to a voisp) numbers, which can be registered in enum? Then you could use those numbers for extensions. But they would only be callable by enum. If the calling of extensions is only to be used by knowledgeable friends you could have them add your own enum-domain to their setup. Leif Hi Leif! No, I cannot believe that this was the right way. It would mean that I would have to register ( pay !!) for every single extension. Just as you would have to pay for a bunch of PSTN-numbers if people should be able to call into the extensions via PSTN As ENUM is implemented, it is a mapping of PSTN-numbers to routing. There is not an option to further delegate numbers below a PSTN-number to extensions. But people can dial into your Asterisk via ENUM, and then dial the extension at the voice prompt. However, at e164.org, you can get a FREE164 number out of the +882 99 number pool. These numbers are for dialing Internet hosts only, they are not 'real' telephone numbers! There you can get a series like 88299 008971 0 to 88299 008971 for your extensions. Leif ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP over TCP/TLS for 1.4 branch
On Mon, Nov 23, 2009 at 4:05 PM, Ekelund, Bryan bekel...@whisolutions.com wrote: Any word on when (or if) SIP over TCP for 1.4 branch is making an appearance? Looking to possibly do an OCS integration, but would prefer to not upgrade to 1.6 or throw OpenSer/Kamailio in the mix. Just file a bug with Microsoft and ask them to support SIP over UDP. Problem solved. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP over TCP/TLS for 1.4 branch
And take the easy way out? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Backeberg Sent: Monday, November 23, 2009 4:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP over TCP/TLS for 1.4 branch On Mon, Nov 23, 2009 at 4:05 PM, Ekelund, Bryan bekel...@whisolutions.com wrote: Any word on when (or if) SIP over TCP for 1.4 branch is making an appearance? Looking to possibly do an OCS integration, but would prefer to not upgrade to 1.6 or throw OpenSer/Kamailio in the mix. Just file a bug with Microsoft and ask them to support SIP over UDP. Problem solved. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users STATEMENT OF CONFIDENTIALITY: The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify WHI Solutions immediately at g...@whisolutions.com, and destroy all copies of this message and any attachments. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Got SIP response 420 Bad Extension back from inphonex.com
Hello: New to asterisk and hoping to use for http://summitcamp.org research station. While trying to use with Inphonex I find that incoming calls drop after about one minute-- -- Got SIP response 420 Bad Extension back from 208.239.76.169 == Spawn extension (incoming-inphonex, 210, 1) exited non-zero on 'SIP/inphonex-095bf208' Found that I can use `*CLI sip set debug peer inphonex` to see more information, such as-- --- SIP read from UDP://208.239.76.169:5060 --- SIP/2.0 420 Bad Extension Via: SIP/2.0/UDP 64.165.113.66:5060;received=64.165.113.66;branch=z9hG4bK121e8b66;rport=5060 From: sip:s-3sdr5c32d1...@10.0.5.66:5060;useradd=64.165.113.66;userport=5060;transport=udp;tag=as111b0d1e To: Unknown sip:6508592...@67.16.112.165;tag=SDbapb901-2318a5d8 Call-ID: SDbapb901-ff4e360f8a8714144f03eb06aad237b5-gurpkk2 CSeq: 102 INVITE Unsupported: timer Content-Length: 0 The best I can figure is that inphonex does not support session-timers because when I insert the following-- sip.conf |session-timers=refuse The calls do not drop. Question is simply whether this will haunt me elsewhere. Thanks, Andrew Using CentOS release 5.4 (Final) / asterisk16-1.6.0.17-1_centos5 Registered to Inphonex-- register = virtuser:passwd:virtu...@sip.inphonex.com:5060/DID [inphonex] username=xxx type=peer secret=xxx ; password used to login their website (same as in register =) host=sip.inphonex.com fromuser=xxx fromdomain=sip.inphonex.com context=incoming-inphonex ; context to be used in extensions.conf for inbound calls from inphonex canreinvite=no insecure=invite ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] can't get pap2 to register from outside the LAN.
I am having a hell of a problem trying to get a linksys pap2t to register with my asterisk from outside the LAN. I have tried every combination of NAT, outbound proxy, stun, specify external IP address etc and it just won't work. Here are the relevant details. In asterisk I have set the following. externip=my.ip.address localnet=192.168.0.0/255.255.0.0 nat=yes bindport=5060 here is the sip user deny=0.0.0.0/0.0.0.0 type=friend secret=blah qualify=yes port=5060 pickupgroup= permit=0.0.0.0/0.0.0.0 nat=yes mailbox=...@device host=dynamic dtmfmode=rfc2833 dial=SIP/372 context=from-internal canreinvite=no callgroup= callerid=device 372 accountcode= call-limit=50 I have tried nat = no, nat=never, nat=route, and leaving out the nat no difference. On the linksys end I have tried everything I can think of. Nat, no nat, stun, hard coded external IP address etc. I have read dozens of web sites and have tried every suggestion given but no joy. I know other people have had the same problem but none of the links I ran into had a solution that worked for me. This device connects perfectly when inside the lan, take it out and it won't connect no matter what I do. Here is the sip debug trace. What truly puzzles me is the 401 not authorized packets. The password is correct, it connects fine inside the lan but the same username and password fails outside the LAN. [Nov 24 14:18:41] --- Transmitting (NAT) to 218.101.6.157:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.50.183:5060;branch=z9hG4bK-26ca393d;received=218.101.6.157 From: 372 sip:3...@203.109.148.108;tag=e25fccc07a79cd65o0 To: 372 sip:3...@203.109.148.108;tag=as1f31845b Call-ID: f4e6d9bc-59a7c...@192.168.50.183 CSeq: 26779 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=0dc307da Content-Length: 0 [Nov 24 14:18:41] Scheduling destruction of SIP dialog 'f4e6d9bc-59a7c...@192.168.50.183' in 32000 ms (Method: REGISTER) [Nov 24 14:18:42] ip --- SIP read from 218.101.6.157:5060 --- REGISTER sip:203.109.148.108 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.183:5060;branch=z9hG4bK-26ca393d From: 372 sip:3...@203.109.148.108;tag=e25fccc07a79cd65o0 To: 372 sip:3...@203.109.148.108 Call-ID: f4e6d9bc-59a7c...@192.168.50.183 CSeq: 26779 REGISTER Max-Forwards: 70 Contact: 372 sip:3...@192.168.50.183:5060;expires=3600 User-Agent: Linksys/PAP2T-5.1.6(LS) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces - [Nov 24 14:18:42] --- (12 headers 0 lines) --- [Nov 24 14:18:42] Using latest REGISTER request as basis request [Nov 24 14:18:42] Sending to 218.101.6.157 : 5060 (NAT) [Nov 24 14:18:42] --- Transmitting (NAT) to 218.101.6.157:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.50.183:5060;branch=z9hG4bK-26ca393d;received=218.101.6.157 From: 372 sip:3...@203.109.148.108;tag=e25fccc07a79cd65o0 To: 372 sip:3...@203.109.148.108 Call-ID: f4e6d9bc-59a7c...@192.168.50.183 CSeq: 26779 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:3...@203.109.148.108 Content-Length: 0 [Nov 24 14:18:42] --- Transmitting (NAT) to 218.101.6.157:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.50.183:5060;branch=z9hG4bK-26ca393d;received=218.101.6.157 From: 372 sip:3...@203.109.148.108;tag=e25fccc07a79cd65o0 To: 372 sip:3...@203.109.148.108;tag=as1f31845b Call-ID: f4e6d9bc-59a7c...@192.168.50.183 CSeq: 26779 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=0dc307da Content-Length: 0 [Nov 24 14:18:42] Scheduling destruction of SIP dialog 'f4e6d9bc-59a7c...@192.168.50.183' in 32000 ms (Method: REGISTER) [Nov 24 14:18:44] ip --- SIP read from 218.101.6.157:5060 --- REGISTER sip:203.109.148.108 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.183:5060;branch=z9hG4bK-26ca393d From: 372 sip:3...@203.109.148.108;tag=e25fccc07a79cd65o0 To: 372 sip:3...@203.109.148.108 Call-ID: f4e6d9bc-59a7c...@192.168.50.183 CSeq: 26779 REGISTER Max-Forwards: 70 Contact: 372 sip:3...@192.168.50.183:5060;expires=3600 User-Agent: Linksys/PAP2T-5.1.6(LS) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces - [Nov 24 14:18:44] --- (12 headers 0 lines) --- [Nov 24 14:18:44] Using latest REGISTER request as basis request [Nov 24 14:18:44] Sending to 218.101.6.157 : 5060 (NAT) [Nov 24 14:18:44] --- Transmitting (NAT) to 218.101.6.157:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.50.183:5060;branch=z9hG4bK-26ca393d;received=218.101.6.157 From: 372 sip:3...@203.109.148.108;tag=e25fccc07a79cd65o0 To: 372 sip:3...@203.109.148.108 Call-ID:
Re: [asterisk-users] can't get pap2 to register from outside the LAN.
I would without the deny and permit directives in the SIP, and rule out some sort of clash there that is rejecting the address the registration is coming from, and take it from there. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Uckun Sent: Tuesday, 24 November 2009 12:26 To: asterisk-users@lists.digium.com Subject: [asterisk-users] can't get pap2 to register from outside the LAN. I am having a hell of a problem trying to get a linksys pap2t to register with my asterisk from outside the LAN. I have tried every combination of NAT, outbound proxy, stun, specify external IP address etc and it just won't work. Here are the relevant details. In asterisk I have set the following. externip=my.ip.address localnet=192.168.0.0/255.255.0.0 nat=yes bindport=5060 here is the sip user deny=0.0.0.0/0.0.0.0 type=friend secret=blah qualify=yes port=5060 pickupgroup= permit=0.0.0.0/0.0.0.0 nat=yes mailbox=...@device host=dynamic dtmfmode=rfc2833 dial=SIP/372 context=from-internal canreinvite=no callgroup= callerid=device 372 accountcode= call-limit=50 I have tried nat = no, nat=never, nat=route, and leaving out the nat no difference. On the linksys end I have tried everything I can think of. Nat, no nat, stun, hard coded external IP address etc. I have read dozens of web sites and have tried every suggestion given but no joy. I know other people have had the same problem but none of the links I ran into had a solution that worked for me. This device connects perfectly when inside the lan, take it out and it won't connect no matter what I do. Here is the sip debug trace. What truly puzzles me is the 401 not authorized packets. The password is correct, it connects fine inside the lan but the same username and password fails outside the LAN. [Nov 24 14:18:41] --- Transmitting (NAT) to 218.101.6.157:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.50.183:5060;branch=z9hG4bK-26ca393d;received=218.101.6.157 From: 372 sip:3...@203.109.148.108;tag=e25fccc07a79cd65o0 To: 372 sip:3...@203.109.148.108;tag=as1f31845b Call-ID: f4e6d9bc-59a7c...@192.168.50.183 CSeq: 26779 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=0dc307da Content-Length: 0 [Nov 24 14:18:41] Scheduling destruction of SIP dialog 'f4e6d9bc-59a7c...@192.168.50.183' in 32000 ms (Method: REGISTER) [Nov 24 14:18:42] ip --- SIP read from 218.101.6.157:5060 --- REGISTER sip:203.109.148.108 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.183:5060;branch=z9hG4bK-26ca393d From: 372 sip:3...@203.109.148.108;tag=e25fccc07a79cd65o0 To: 372 sip:3...@203.109.148.108 Call-ID: f4e6d9bc-59a7c...@192.168.50.183 CSeq: 26779 REGISTER Max-Forwards: 70 Contact: 372 sip:3...@192.168.50.183:5060;expires=3600 User-Agent: Linksys/PAP2T-5.1.6(LS) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces - [Nov 24 14:18:42] --- (12 headers 0 lines) --- [Nov 24 14:18:42] Using latest REGISTER request as basis request [Nov 24 14:18:42] Sending to 218.101.6.157 : 5060 (NAT) [Nov 24 14:18:42] --- Transmitting (NAT) to 218.101.6.157:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.50.183:5060;branch=z9hG4bK-26ca393d;received=218.101.6.157 From: 372 sip:3...@203.109.148.108;tag=e25fccc07a79cd65o0 To: 372 sip:3...@203.109.148.108 Call-ID: f4e6d9bc-59a7c...@192.168.50.183 CSeq: 26779 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:3...@203.109.148.108 Content-Length: 0 [Nov 24 14:18:42] --- Transmitting (NAT) to 218.101.6.157:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.50.183:5060;branch=z9hG4bK-26ca393d;received=218.101.6.157 From: 372 sip:3...@203.109.148.108;tag=e25fccc07a79cd65o0 To: 372 sip:3...@203.109.148.108;tag=as1f31845b Call-ID: f4e6d9bc-59a7c...@192.168.50.183 CSeq: 26779 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=0dc307da Content-Length: 0 [Nov 24 14:18:42] Scheduling destruction of SIP dialog 'f4e6d9bc-59a7c...@192.168.50.183' in 32000 ms (Method: REGISTER) [Nov 24 14:18:44] ip --- SIP read from 218.101.6.157:5060 --- REGISTER sip:203.109.148.108 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.183:5060;branch=z9hG4bK-26ca393d From: 372 sip:3...@203.109.148.108;tag=e25fccc07a79cd65o0 To: 372 sip:3...@203.109.148.108 Call-ID: f4e6d9bc-59a7c...@192.168.50.183 CSeq: 26779 REGISTER Max-Forwards: 70 Contact: 372 sip:3...@192.168.50.183:5060;expires=3600 User-Agent: Linksys/PAP2T-5.1.6(LS) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces
[asterisk-users] DIDs PBX Multi-channel balanced audio output?
There came up an interesting question on last weeks VoIP Users Conf call. http://vuc.me. An internet broadcaster wanted to be able to take a number of phone calls and bring them into a traditional audio mixer where each call was a separate balanced audio source to the mixer. Let's say that he might need 8 simultaneous calls. Is there a way to set this up using a single Asterisk server and the monitor process to send the various call streams to a multi-channel audio interface card? He wants to mix them external to the server in the audio console so there's no MeetMe involved. It's perhaps more obvious to suggest using small format hardware (ALIX?) and a USB audio interface to effect a single line interface. But then extending to more tha n afew lines might get unwieldy. Any ideas? Michael -- Michael Graves mgravesatmstvp.com http://www.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves Twitter mjgraves ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DIDs PBX Multi-channel balanced audio output?
Hi! Is there a way to set this up using a single Asterisk server and the monitor process to send the various call streams to a multi-channel audio interface card? He wants Any ideas? Jackaudio? That would require 1.6. http://www.voip-info.org/wiki/view/Asterisk+cmd+jack Philipp ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] can't get pap2 to register from outside the LAN.
How about adding: insecure=invite,port --- On Mon, 11/23/09, Tim Uckun timuc...@gmail.com wrote: From: Tim Uckun timuc...@gmail.com Subject: [asterisk-users] can't get pap2 to register from outside the LAN. To: asterisk-users@lists.digium.com Date: Monday, November 23, 2009, 8:25 PM I am having a hell of a problem trying to get a linksys pap2t to register with my asterisk from outside the LAN. I have tried every combination of NAT, outbound proxy, stun, specify external IP address etc and it just won't work. Here are the relevant details. In asterisk I have set the following. externip=my.ip.address localnet=192.168.0.0/255.255.0.0 nat=yes bindport=5060 here is the sip user deny=0.0.0.0/0.0.0.0 type=friend secret=blah qualify=yes port=5060 pickupgroup= permit=0.0.0.0/0.0.0.0 nat=yes mailbox=...@device host=dynamic dtmfmode=rfc2833 dial=SIP/372 context=from-internal canreinvite=no callgroup= callerid=device 372 accountcode= call-limit=50 I have tried nat = no, nat=never, nat=route, and leaving out the nat no difference. On the linksys end I have tried everything I can think of. Nat, no nat, stun, hard coded external IP address etc. I have read dozens of web sites and have tried every suggestion given but no joy. I know other people have had the same problem but none of the links I ran into had a solution that worked for me. This device connects perfectly when inside the lan, take it out and it won't connect no matter what I do. Here is the sip debug trace. What truly puzzles me is the 401 not authorized packets. The password is correct, it connects fine inside the lan but the same username and password fails outside the LAN. [Nov 24 14:18:41] --- Transmitting (NAT) to 218.101.6.157:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.50.183:5060;branch=z9hG4bK-26ca393d;received=218.101.6.157 From: 372 sip:3...@203.109.148.108;tag=e25fccc07a79cd65o0 To: 372 sip:3...@203.109.148.108;tag=as1f31845b Call-ID: f4e6d9bc-59a7c...@192.168.50.183 CSeq: 26779 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=0dc307da Content-Length: 0 [Nov 24 14:18:41] Scheduling destruction of SIP dialog 'f4e6d9bc-59a7c...@192.168.50.183' in 32000 ms (Method: REGISTER) [Nov 24 14:18:42] ip --- SIP read from 218.101.6.157:5060 --- REGISTER sip:203.109.148.108 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.183:5060;branch=z9hG4bK-26ca393d From: 372 sip:3...@203.109.148.108;tag=e25fccc07a79cd65o0 To: 372 sip:3...@203.109.148.108 Call-ID: f4e6d9bc-59a7c...@192.168.50.183 CSeq: 26779 REGISTER Max-Forwards: 70 Contact: 372 sip:3...@192.168.50.183:5060;expires=3600 User-Agent: Linksys/PAP2T-5.1.6(LS) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces - [Nov 24 14:18:42] --- (12 headers 0 lines) --- [Nov 24 14:18:42] Using latest REGISTER request as basis request [Nov 24 14:18:42] Sending to 218.101.6.157 : 5060 (NAT) [Nov 24 14:18:42] --- Transmitting (NAT) to 218.101.6.157:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.50.183:5060;branch=z9hG4bK-26ca393d;received=218.101.6.157 From: 372 sip:3...@203.109.148.108;tag=e25fccc07a79cd65o0 To: 372 sip:3...@203.109.148.108 Call-ID: f4e6d9bc-59a7c...@192.168.50.183 CSeq: 26779 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:3...@203.109.148.108 Content-Length: 0 [Nov 24 14:18:42] --- Transmitting (NAT) to 218.101.6.157:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.50.183:5060;branch=z9hG4bK-26ca393d;received=218.101.6.157 From: 372 sip:3...@203.109.148.108;tag=e25fccc07a79cd65o0 To: 372 sip:3...@203.109.148.108;tag=as1f31845b Call-ID: f4e6d9bc-59a7c...@192.168.50.183 CSeq: 26779 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=0dc307da Content-Length: 0 [Nov 24 14:18:42] Scheduling destruction of SIP dialog 'f4e6d9bc-59a7c...@192.168.50.183' in 32000 ms (Method: REGISTER) [Nov 24 14:18:44] ip --- SIP read from 218.101.6.157:5060 --- REGISTER sip:203.109.148.108 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.183:5060;branch=z9hG4bK-26ca393d From: 372 sip:3...@203.109.148.108;tag=e25fccc07a79cd65o0 To: 372 sip:3...@203.109.148.108 Call-ID: f4e6d9bc-59a7c...@192.168.50.183 CSeq: 26779 REGISTER Max-Forwards: 70 Contact: 372 sip:3...@192.168.50.183:5060;expires=3600 User-Agent: Linksys/PAP2T-5.1.6(LS) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces
[asterisk-users] asterisk trunk CURL hangs in the dialplan
We've encountered a strange issue with the trunk version of asterisk.
Our dialplan makes CURL calls and occasionally CURL stops working.
The dialplan looks something like this:
[macro-curl]
; ${ARG1} CURL URL
; ${ARG2} CURL POST
exten = s,1,NoOp(CURL)
...
exten =
s,n(post),Set(RF_CURL_POST=userID=${RF_DIALER_USERID}password=${RF_PASSWORD}${ARG2})
exten = s,n,Set(CURLOPT(httptimeout)=5)
exten = s,n,Set(CURLOPT(conntimeout)=5)
exten = s,n,NoOp(CURL(${RF_URL}/${ARG1}?${RF_CURL_POST}))
exten = s,n,Set(RF_CURL_RESPONSE=${CURL(${RF_URL}/${ARG1},${RF_CURL_POST})})
At this point, CURL either works or it will occasionally hang for a few
minutes. tcpdump doesn't show any traffic from the asterisk box to the web
server.
Something seems to be causing CURL to hang, before it sends out the http
request and the CURLOPT timeouts have no effect on the behavior.
Once CURL hangs, any additional calls to CURL also hang.
After a few minutes, tcpdump will show the CURL traffic going to the web server.
And CURL begins functioning normally for a while.
Has anyone else seen this? Or have any suggestions on how to debug this?
--
Eric Chamberlain, Founder
RF.com - http://RF.com/
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