[asterisk-users] chan_ss7 with ringing, but no voice stream.
hello, all of users:
sorry, resend it again for clarifying the message. I have implemented cha_ss7
in china. initially, the
chan_ss7 can not support the call group. i modify the code.
now the problem is that, both sides can hear the ring, but i
can not hear the voice from each other.
i think the ss7 does not send the voice steam to the destination.
in chan_ss7, i added:
===
static struct ss7_chan *cic_hunt_even_mru(struct linkset*
linkset) {
struct ss7_chan *cur, *prev, *best, *best_prev;
best = NULL;
best_prev = NULL;
for(cur = linkset-idle_list, prev = NULL; cur !=
NULL; prev = cur, cur = cur-next_idle) {
/* Don't select lines that are resetting or
blocked. */
if(!cur-reset_done || (cur-blocked
(BL_LH|BL_RM|BL_RH|BL_UNEQUIPPED|BL_LINKDOWN))) {
continue;
}
/* if((cur-cic % 2) == 0) { */
/*change to this*/
if(((cur-cic % 2) ==
0)0==strcasecmp(cur-link-name,linkname))
{
/* Choose the first idle even circuit,
if any. */
/*end of change*/
best = cur;
best_prev = prev;
break;
} else if(best == NULL) {
/* Remember the first odd circuit, in
case no even circuits are
available. */
best = cur;
best_prev = prev;
}
}
cic_hunt_even_mru if(((cur-cic % 2) ==
0)0==strcasecmp(cur-link-name,linkname))
{
my environment is:
asterisk-1.4.20
chan_ss7-1.0.91
Openvox D410P
===
anyone has an idea for the problem?
please give me some hints!
thanks!
james.zhu
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Re: [asterisk-users] Magic SIP Phone
On Thu, 19 Mar 2009, Christian Victor wrote: grandstream gxp-2000 works fine for that. depending on firmware rev its two ports are either a switch or router. Grandstream removed this functionality in recent softwware upgrades - I guess they needed the code space for other things. Why would you want a router in the phone and not let the PC connected to the phones internal (mini) switch get an IP from the DHCP server in the cable box? Or will that deliver only ONE IP at a time? Most cable modems are jsut that - a dumb media convertor. The DHCP server back at HQ was set to give just one IP address per subcriber - at least in the UK that was how it was.(still is I think) The routers inside the phones were relatively full fetured IIRC, offering DHCP, NAT, port forwarding, etc. I suspect they were intended mainly for the US market where cable was the main technology at the time - and presented via Ethernet off the modem. Didn't mainland europe embrace ADSL from the start? The UK was cable in the early days but then BT got wise and we have near 100% ADSL avalability, as well as the existing cable stuff, but everyone I know with cable went out bought a router anyway... (Telewest, etc. now Virgin Media) were a bit restrictive in the number of PCs you could connect, etc.) Gordon ___ -- 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] Special Information Tones
On Fri, 20 Mar 2009, Stephen Davies wrote: Hi, Are you sure that Verizon amswers the call? They should play that message as 'early media' without answering, after which they cam clear the call with an appropriate cause code. Similar issue in the UK and yes, the carriers do answer the call - because from that second onward thy are taking revenue. BT offer a free voicemailbox on landlines too - for the same reason. Gordon That would work for you and still give callers the audible ,essage they want. Steve On 3/20/09, drew einhorn drew.einh...@gmail.com wrote: I'm having a problem with Verizon Wireless. I would be extremely surprised if I was the only one having this problem. It seems to me that Verizon Wireless might be able to use one of the Special Information Tones to allow us to solve the problem. But I really do not whether my suggestion is compliant with the ITU-T standards. Perhaps someone can give me an expert opinion on whether I should try to get Verizon to implement my suggestion. First I'll describe the problem. I'm trying to implement Single Number Reach. For example, when a call comes in to one of my DIDs, it simultaneously rings on a couple extensions in my home office and a couple of Verizon Wireless cell phone numbers. Everything works just the way it is supposed to if the cell phones are powered up, and within the range of a cell tower. The problem is if a cellphone is turned off, or out of range and unable to talk to a cell tower, Verizon is unable to find the cellphone on their network, Verizon answers the call and plays a recorded message, instead of allowing the number to continue ringing, and allowing one of the voip extensions, or another cellphone to answer the call. Verizon really wants to get rid of the call as quickly as possible to free up their equipment to handle other calls. Unfortunately we spend a lot of time in rural areas where there is no cell tower to talk to. In that case we really someone else to pick up the call. I'm hoping that if Verizon would precede the voice message with one of the Special Information Tones, we could recognize the fact that the call has not really been answer, and continue to ring on the other lines. Two questions. 1) would the approach be compliant with ITU-T standards? 2) Assuming that it is, and we can convince Verizon to implement this. How difficult would it be to configure asterisk to handle this as I suggest? -- Drew Einhorn ___ -- 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 -- Sent from my mobile device ___ -- 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] Hardware suggestions
On Thu, 19 Mar 2009 16:38:02 -0300, David fire ddf...@gmail.com wrote: dive in the mailing list archive in February a very nice user sent an email about how to do load balancing using opensip. I don't suppose you know the Subject line, do you David? I can't find it! ___ -- 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] Asterisk and PBX internal numbers
2009/3/20 D Tucny d...@tucny.com
2009/3/19 Oguzhan Kayhan oguzh...@bilkent.edu.tr
Hi, i know i am asking a lot of questions lately in this forum..sorry
about that first of all. :)
Ok, here is the deal..
I am trying to make a hybrid system with an ericsson MD110 and
asterisk.
Internally we have 4 digit phone extensions on ericsson.. and so in
asterisk.
So, what i want to do is to call pbx side without adding 9 or etc to
the
begining of the number from asterisk clients..
For example if smbody tries to call 1500 first asterisk should look for
its local extensions and, if it not there it should dial PRI trunk..
Is there any way for that?
It's pretty simple... e.g.
[context-for-phones]
include = localphones
include = everythingelse
[localphones]
exten = 1500,1,Dial(SIP/1500)
exten = 1501,1,Dial(SIP/1501)
exten = 1502,1,Dial(SIP/1502)
[everythingelse]
exten = ,1,Dial(DAHDI/g1/${EXTEN})
So, your phones would be set up to use context-for-phones, then if a
call
was made the localphones context would be checked for a match before the
everythingelse context... If 1500/1501/1502 were dialled, the calls
would go
the SIP/1500/1/2, any other 4 digit extension would be sent to the DAHDI
group 1...
Yes seems pretty simple,
so can i use a wildcard instead of giving all numbers one by one?
Such
instead of exten = 1500,1,Dial(SIP/1500)
exten = _15XX,1,Dial(SIP/_15XX) ?
assuming all my local numbers are on 15 prefix??
Sorry, small correction my example is missing a '_' before which
would
be needed...
d
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[asterisk-users] Max concurrent calls
Hi all, I mentioned in asterisk.conf there is a property maxcalls...I know that this is the max number of concurrent calls but i need to know please if this entry is commented out, what is the default number of MAX concurrent calls supported by asterisk? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Realtime Configuration and 404 Extension not found
Hi to all the ML. I'm new here.
I start to use asterisk with realtime configuration, with pgsql
backend connected via odbc.
The connection between asterisk and pgsql works fine.
I create a table sip_conf with 2 user (for testing purpose), 1401 and 1501.
Those are the records:
asterisk= SELECT name,host,type,context,secret,defaultuser from sip_conf;
name | host | type | context | secret | defaultuser
--+-++-++-
1401 | dynamic | friend | prova | fra| 1401
1501 | dynamic | friend | prova | 1501 | 1501
I create a table extensions_conf with this extensions:
id | context | exten | priority | app |appdata
+-+---+--++
41 | prova | _1[1-5]XX |1 | Dial | (SIP/${EXTEN})
42 | prova | _1[1-5]XX |2 | Hangup |
the extconfig.conf is:
[settings]
extensions = odbc,dbasterisk,extensions_conf
sipusers = odbc,dbasterisk,sip_conf
sippeers = odbc,dbasterisk,sip_conf
and extensions.conf is:
[general]
static=yes
writeprotect=no
autofallthrough=yes
[prova]
switch = Realtime
I use x-lite for calling 1401 - 1501 and vice-versa.
The result is 404 Not Found. The phones correctly register on
asterisk, and the table sip_conf register the userage, the ip ecc.
Setting sip set debug on on asterisk console, i obtain:
--8---8---8---8---8-
Via: SIP/2.0/UDP
10.44.3.153:5060;rport;branch=z9hG4bK5591DE8FE488A6D64C4AA159B97BBC36
From: 1501 sip:1...@10.44.9.0;tag=990832587
To: sip:1...@10.44.9.0
Contact: sip:1...@10.44.3.153:5060
Call-ID: 448afe95-423a-a081-85f3-ad0d70d0f...@10.44.3.153
CSeq: 60784 INVITE
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1105d
Content-Length: 305
v=0
o=1501 503728492 503728510 IN IP4 10.44.3.153
s=X-Lite
c=IN IP4 10.44.3.153
t=0 0
m=audio 8000 RTP/AVP 0 8 3 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
-
--- (11 headers 14 lines) ---
== Using SIP RTP CoS mark 5
Sending to 10.44.3.153 : 5060 (NAT)
Using INVITE request as basis request -
448afe95-423a-a081-85f3-ad0d70d0f...@10.44.3.153
Found peer '1501' for '1501' from 10.44.3.153:5060
--- Reliably Transmitting (no NAT) to 10.44.3.153:5060 ---
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
10.44.3.153:5060;branch=z9hG4bK5591DE8FE488A6D64C4AA159B97BBC36;received=10.44.3.153;rport=5060
From: 1501 sip:1...@10.44.9.0;tag=990832587
To: sip:1...@10.44.9.0;tag=as4b3729a3
Call-ID: 448afe95-423a-a081-85f3-ad0d70d0f...@10.44.3.153
CSeq: 60784 INVITE
Server: Asterisk PBX 1.6.1-rc1-issue14292
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=10.44.9.0,
nonce=4685095b
Content-Length: 0
Scheduling destruction of SIP dialog
'448afe95-423a-a081-85f3-ad0d70d0f...@10.44.3.153' in 32000 ms
(Method: INVITE)
itpbx01*CLI
--- SIP read from UDP:10.44.3.153:5060 ---
ACK sip:1...@10.44.9.0 SIP/2.0
Via: SIP/2.0/UDP
10.44.3.153:5060;rport;branch=z9hG4bK5591DE8FE488A6D64C4AA159B97BBC36
From: 1501 sip:1...@10.44.9.0;tag=990832587
To: sip:1...@10.44.9.0;tag=as4b3729a3
Contact: sip:1...@10.44.3.153:5060
Call-ID: 448afe95-423a-a081-85f3-ad0d70d0f...@10.44.3.153
CSeq: 60784 ACK
Max-Forwards: 70
Content-Length: 0
-
--- (9 headers 0 lines) ---
itpbx01*CLI
--- SIP read from UDP:10.44.3.153:5060 ---
INVITE sip:1...@10.44.9.0 SIP/2.0
Via: SIP/2.0/UDP
10.44.3.153:5060;rport;branch=z9hG4bK26FAB8569057F005F91285F58781AD32
From: 1501 sip:1...@10.44.9.0;tag=990832587
To: sip:1...@10.44.9.0
Contact: sip:1...@10.44.3.153:5060
Call-ID: 448afe95-423a-a081-85f3-ad0d70d0f...@10.44.3.153
CSeq: 60785 INVITE
Authorization: Digest
username=1501,realm=10.44.9.0,nonce=4685095b,response=290748094c61098ad389dafa825e,uri=sip:1...@10.44.9.0,algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1105d
Content-Length: 305
v=0
o=1501 503728492 503728510 IN IP4 10.44.3.153
s=X-Lite
c=IN IP4 10.44.3.153
t=0 0
m=audio 8000 RTP/AVP 0 8 3 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
-
--- (12 headers 14 lines) ---
Sending to 10.44.3.153 : 5060 (NAT)
Using INVITE request as basis request -
448afe95-423a-a081-85f3-ad0d70d0f...@10.44.3.153
Found peer '1501' for '1501' from 10.44.3.153:5060
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 98
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 10.44.3.153:8000
Found audio description format pcmu for ID 0
Found audio description format pcma for ID 8
Found audio description
[asterisk-users] T38 FAX
Dear All, I'm trying to send FAX to an endpoint Behind NAT...The scenario i the following: PSTN_GW--Asterisk--asterisk--OpenSIPS--Endpoint behind NAT.. The FAX is failed and I got the following error log on asterisk: Mar 20 09:21:09] WARNING[20388]: chan_sip.c:12409 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog '76c64f8373def68d5088690816c67...@asterisk_ip'. Giving up. Please find the whole trace at http://pastebin.com/d764514b4 Can you please help me in order to find the real issue? ___ -- 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] log to cdr each dialpan action, not only one record for each call
On Fri, 13 Mar 2009 09:22:12 +0100, Lenz Emilitri wrote: I'm only half joking: what about parsing the full log looking for command inviocations and channel IDs? this would be completely transparent, albeit insane :) The full log is insane on a busy server. You get no decent call tracing, you depend on the format of that log to not change, parsing every line is _expensive_, and there are issues with log rotation. Following the AMI events, on the other hand, is easy to set up, allows for discriminating between calls in a reasonably sane manner, doesn't have data format problems, parses quickly, and is a simple TCP data stream that can even be accessed remotely. -- ___ -- 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] Direct Dial-Out and CDR destination numbers
On Tue, 17 Mar 2009 13:34:34 +, Geraint Lee wrote:
what about relogging the information using:
Set(CDR(customfield)=${CDR(originalfield)})
That won't work because there's no way (that I know of) to collect the
DTMF numbers which the Dial() command consumes in order to complete
the destination number.
One workaround would be to play with dial patterns and timeouts.
But I want to avoid that, as (a) it can cause digits to be lost if
they're dialled at exactly the wrong time and (b) I want my calls to
complete as soon as possible -- a two-second let's wait if the user
seems to be done before trying to dial delay is not acceptable.
--
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Re: [asterisk-users] T.38 - Which endpoint shall reINVITE ? caller or callee ?
Steve Underwood schrieb: Hi Olivier, Olivier wrote: T.38 says that if the call starts in audio mode it is the called end which should initiate a re-invite to change from audio to T.38. This makes sense, as that is the end which has the best chance of figuring out if a FAX machine answers the call. In practice many T.38 implementations will send out a re-invite when they are the calling side, so any practical implementation has to allow for this. Clashes are possible, if both ends send re-invite, and this is not always handled properly Yesterday, with 2 consecutive sendings on the same setup (same fax file, same ATAs, same servers), on the first try, I've seen the reINVITE coming from callee on from the caller on the second try. I don't remember I changed anything between both tries (though I may have done without noticing this). That is what typically what happens when the calling end doesn't obey the spec. It comes down to a race for who initiates the re-invite first. If you are lucky the two ends sort themselves out. If you are unlucky you end up with both ends re-inviting, and you may get a call failure. This is what I see very often - both sides send reINVITE (overlapping), both sides reject with 491 (request pending) with the result that the switch to T.38 failed. I have only once seen handling overlapping reINVITEs smart (ComISDN client) Theoretically it would be perfect if the reINVITE is always triggered by the callee (as the spec says) - but there are ATAs which need up to 15 seconds to detect a fax and send reINVITE - and in this cases you sometimes have to work around by allowing the caller to send reINVITE too. regards Klaus ___ -- 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] VM_DATE in french?
Configure emaildateformat in voicemail.conf.
I worked around the english weekdays by using numeric weekdays (see man
strftime)
emaildateformat=%d. %m. %Y um %H:%M Uhr
If you need the weekday in French you have to set the Linux Locale to
french. But this affects all parts of Asterisk where timestamps are
generated, e.g. CDRs ...
You can also see the discussion on:
http://bugs.digium.com/view.php?id=14333
regards
klaus
BERGANZ François schrieb:
Hello,
I work on voicemail.conf and I need that ${VM_DATE} is in french!
How can I do it?
Thank you
Cordialement,
BERGANZ François
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
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Re: [asterisk-users] Asterisk and PBX internal numbers
Quoting Oguzhan Kayhan oguzh...@bilkent.edu.tr:
Yes seems pretty simple,
so can i use a wildcard instead of giving all numbers one by one?
Such
instead of exten = 1500,1,Dial(SIP/1500)
exten = _15XX,1,Dial(SIP/_15XX) ?
assuming all my local numbers are on 15 prefix??
Almost.
exten = _15XX,1,Dial(SIP/${EXTEN})
--
Phil Reynolds
mail: phil-aster...@tinsleyviaduct.com
Web: http://www.tinsleyviaduct.com/phil/
Waltham 66, Emley Moor 69, Droitwich 79, Windows 95
This message was sent using IMP, the Internet Messaging Program.
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Re: [asterisk-users] chan_ss7 with ringing, but no voice stream.
SS7 doesnt send any voice. It sends call info, and tells the switches
which trunk to use for the voice. Trunks are two-way as far as audio
content, though they maybe designated is inbound or outbound trunks.
An audio problem is possibly a NAT or other issue.
Since you are modifying the SS7 code, there could be some error in setting
up the call, but normally the IMT trunks are two way. (Of course they are 4
wire circuits so are two one way paths, but they are matched pairs so,
for practical purposes they would be 1 entity for call set up purposes.)
Cary Fitch
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of lizhong zhu
Sent: Friday, March 20, 2009 2:05 AM
To: asterisk-ss7
Subject: [asterisk-users] chan_ss7 with ringing, but no voice stream.
hello, all of users:
sorry, resend it again for clarifying the message. I have implemented
cha_ss7 in china. initially, the
chan_ss7 can not support the call group. i modify the code.
now the problem is that, both sides can hear the ring, but i
can not hear the voice from each other.
i think the ss7 does not send the voice steam to the destination.
in chan_ss7, i added:
===
static struct ss7_chan *cic_hunt_even_mru(struct linkset*
linkset) {
struct ss7_chan *cur, *prev, *best, *best_prev;
best = NULL;
best_prev = NULL;
for(cur = linkset-idle_list, prev = NULL; cur !=
NULL; prev = cur, cur = cur-next_idle) {
/* Don't select lines that are resetting or
blocked. */
if(!cur-reset_done || (cur-blocked
(BL_LH|BL_RM|BL_RH|BL_UNEQUIPPED|BL_LINKDOWN))) {
continue;
}
/* if((cur-cic % 2) == 0) { */
/*change to this*/
if(((cur-cic % 2) ==
0)0==strcasecmp(cur-link-name,linkname))
{
/* Choose the first idle even circuit,
if any. */
/*end of change*/
best = cur;
best_prev = prev;
break;
} else if(best == NULL) {
/* Remember the first odd circuit, in
case no even circuits are
available. */
best = cur;
best_prev = prev;
}
}
cic_hunt_even_mru if(((cur-cic % 2) ==
0)0==strcasecmp(cur-link-name,linkname))
{
my environment is:
asterisk-1.4.20
chan_ss7-1.0.91
Openvox D410P
===
anyone has an idea for the problem?
please give me some hints!
thanks!
james.zhu
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Re: [asterisk-users] Special Information Tones
On Fri, Mar 20, 2009 at 1:53 AM, Gordon Henderson gordon+aster...@drogon.net wrote: On Fri, 20 Mar 2009, Stephen Davies wrote: Hi, Are you sure that Verizon amswers the call? They should play that message as 'early media' without answering, after which they cam clear the call with an appropriate cause code. Yes, They are answering the call, sometimes on the first ring, and taking it away from the lines that should be answering the call. Similar issue in the UK and yes, the carriers do answer the call - because from that second onward thy are taking revenue. BT offer a free voicemailbox on landlines too - for the same reason. So, they really want to answer the phone so they can charge for the call. If we can get them to put one of the Special Information Tones in front of the call, can we make asterisk ignore that false answer and allow the other lines to continue simultaneously ringing until we get a real answer, or it goes to voicemail? Gordon That would work for you and still give callers the audible ,essage they want. Steve On 3/20/09, drew einhorn drew.einh...@gmail.com wrote: I'm having a problem with Verizon Wireless. I would be extremely surprised if I was the only one having this problem. It seems to me that Verizon Wireless might be able to use one of the Special Information Tones to allow us to solve the problem. But I really do not whether my suggestion is compliant with the ITU-T standards. Perhaps someone can give me an expert opinion on whether I should try to get Verizon to implement my suggestion. First I'll describe the problem. I'm trying to implement Single Number Reach. For example, when a call comes in to one of my DIDs, it simultaneously rings on a couple extensions in my home office and a couple of Verizon Wireless cell phone numbers. Everything works just the way it is supposed to if the cell phones are powered up, and within the range of a cell tower. The problem is if a cellphone is turned off, or out of range and unable to talk to a cell tower, Verizon is unable to find the cellphone on their network, Verizon answers the call and plays a recorded message, instead of allowing the number to continue ringing, and allowing one of the voip extensions, or another cellphone to answer the call. Verizon really wants to get rid of the call as quickly as possible to free up their equipment to handle other calls. Unfortunately we spend a lot of time in rural areas where there is no cell tower to talk to. In that case we really someone else to pick up the call. I'm hoping that if Verizon would precede the voice message with one of the Special Information Tones, we could recognize the fact that the call has not really been answer, and continue to ring on the other lines. Two questions. 1) would the approach be compliant with ITU-T standards? 2) Assuming that it is, and we can convince Verizon to implement this. How difficult would it be to configure asterisk to handle this as I suggest? -- Drew Einhorn ___ -- 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 -- Sent from my mobile device ___ -- 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 -- Drew Einhorn ___ -- 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] T38 FAX
On Fri, Mar 20, 2009 at 5:36 AM, michel freiha mich...@gmail.com wrote: Can you please help me in order to find the real issue? Try taking out three or four pieces of your architecture, and then try again. How about PSTN - Asterisk? ___ -- 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] Special Information Tones
From a cell user level perspective... The cell companies are doing it like they think makes sense. If they know your cell is off/out of range they route instantly to VM. They could give 4-10 rings of fake effort, but why. With follow me roaming and such, they want to process the call as fast as possible. If they don't know if the cell is available, they may go through about 4 rings of searching, but beyond that it is time to send it to VM, charge for the call :-), and move on. Ideally, a find me call forwarding system should have a real person identifier and local voice mail. Real person means that all called external numbers should not be assumed to be answered until they send back a DTMF tone. Something like a Background announcement with some silence, waiting for DTMF. It could be a Boing or You have a forwarded call, press any key to accept the call Then the call should be cut through to that extension. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of drew einhorn Sent: Friday, March 20, 2009 8:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Special Information Tones On Fri, Mar 20, 2009 at 1:53 AM, Gordon Henderson gordon+aster...@drogon.net wrote: On Fri, 20 Mar 2009, Stephen Davies wrote: Hi, Are you sure that Verizon amswers the call? They should play that message as 'early media' without answering, after which they cam clear the call with an appropriate cause code. Yes, They are answering the call, sometimes on the first ring, and taking it away from the lines that should be answering the call. Similar issue in the UK and yes, the carriers do answer the call - because from that second onward thy are taking revenue. BT offer a free voicemailbox on landlines too - for the same reason. So, they really want to answer the phone so they can charge for the call. If we can get them to put one of the Special Information Tones in front of the call, can we make asterisk ignore that false answer and allow the other lines to continue simultaneously ringing until we get a real answer, or it goes to voicemail? Gordon That would work for you and still give callers the audible ,essage they want. Steve On 3/20/09, drew einhorn drew.einh...@gmail.com wrote: I'm having a problem with Verizon Wireless. I would be extremely surprised if I was the only one having this problem. It seems to me that Verizon Wireless might be able to use one of the Special Information Tones to allow us to solve the problem. But I really do not whether my suggestion is compliant with the ITU-T standards. Perhaps someone can give me an expert opinion on whether I should try to get Verizon to implement my suggestion. First I'll describe the problem. I'm trying to implement Single Number Reach. For example, when a call comes in to one of my DIDs, it simultaneously rings on a couple extensions in my home office and a couple of Verizon Wireless cell phone numbers. Everything works just the way it is supposed to if the cell phones are powered up, and within the range of a cell tower. The problem is if a cellphone is turned off, or out of range and unable to talk to a cell tower, Verizon is unable to find the cellphone on their network, Verizon answers the call and plays a recorded message, instead of allowing the number to continue ringing, and allowing one of the voip extensions, or another cellphone to answer the call. Verizon really wants to get rid of the call as quickly as possible to free up their equipment to handle other calls. Unfortunately we spend a lot of time in rural areas where there is no cell tower to talk to. In that case we really someone else to pick up the call. I'm hoping that if Verizon would precede the voice message with one of the Special Information Tones, we could recognize the fact that the call has not really been answer, and continue to ring on the other lines. Two questions. 1) would the approach be compliant with ITU-T standards? 2) Assuming that it is, and we can convince Verizon to implement this. How difficult would it be to configure asterisk to handle this as I suggest? -- Drew Einhorn ___ -- 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 -- Sent from my mobile device ___ -- 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
Re: [asterisk-users] T.38 - Which endpoint shall reINVITE ? caller or callee ?
2009/3/20 Klaus Darilion klaus.mailingli...@pernau.at Steve Underwood schrieb: Hi Olivier, Olivier wrote: T.38 says that if the call starts in audio mode it is the called end which should initiate a re-invite to change from audio to T.38. This makes sense, as that is the end which has the best chance of figuring out if a FAX machine answers the call. In practice many T.38 implementations will send out a re-invite when they are the calling side, so any practical implementation has to allow for this. Clashes are possible, if both ends send re-invite, and this is not always handled properly Yesterday, with 2 consecutive sendings on the same setup (same fax file, same ATAs, same servers), on the first try, I've seen the reINVITE coming from callee on from the caller on the second try. I don't remember I changed anything between both tries (though I may have done without noticing this). That is what typically what happens when the calling end doesn't obey the spec. It comes down to a race for who initiates the re-invite first. If you are lucky the two ends sort themselves out. If you are unlucky you end up with both ends re-inviting, and you may get a call failure. This is what I see very often - both sides send reINVITE (overlapping), both sides reject with 491 (request pending) with the result that the switch to T.38 failed. I have only once seen handling overlapping reINVITEs smart (ComISDN client) Theoretically it would be perfect if the reINVITE is always triggered by the callee (as the spec says) - but there are ATAs which need up to 15 seconds to detect a fax and send reINVITE - and in this cases you sometimes have to work around by allowing the caller to send reINVITE too. Linksys 3102 have an option specifying if reINVITE should be inbound, outbound or both. Unfortunately, I couldn't find this option in Patton SmartNodes. regards Klaus ___ -- 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] Asterisk + OpenSIPs Integration - Rewrite URI on Trunk Numbers of a SIP Trunk
Hello All, I have a little complicated question about the Dial command. I use OpenSIPs to loadbalance Asterisk Servers, and Users are registered on Asterisk servers. Asterisk use the Reg. Contact entry to reach the UAC via the OpenSIPs server. Everything works except for trunk numbers: For each peer on Asterisk, Addr-IP is IP of the Proxy and Reg. Contact is the IP where the proxy will relay the packet to reach the UAC. Ex: with a trunk 0123400010 - 0123400019 with 0123400010 as the sip peer. When a number from a trunk is called, like 0123400019 the Reg. Contact of the main number is not used. For the time being, I use Dial(SIP/0123400010/0123400019) but it It sends an INVITE sip:0123400...@proxyip to the proxy whereas it should send INVITE sip:0123400019@Reg. Contact of the main number to the proxy So I'm trying use the Dial Command with Dial(SIP/0123400010/0123400019@Reg. Contact of the main number) but it doesn't work Have you got any idea how to rewrite the IP of the URI sent? Thanks! -- -- -- Marc LEURENT lf...@leurent.eu ___ -- 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] Special Information Tones
On Fri, 20 Mar 2009, Stephen Davies wrote: Hi, Are you sure that Verizon amswers the call? They should play that message as 'early media' without answering, after which they cam clear the call with an appropriate cause code. Similar issue in the UK and yes, the carriers do answer the call - because from that second onward thy are taking revenue. BT offer a free voicemailbox on landlines too - for the same reason. Many carriers allow you to opt out of these sorts of misfeatures, though you may have to be somewhat insistent. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ -- 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] OpenSIPS on CentOS
Hello, I've been looking into OpenSIPS to see if it's a worthwhile addition to our setup. We're currently running a cluster, using Heartbeat, between two servers. It works well but I'm interested in seeing if we can improve it. My manager heavily uses RPM's for installations rather than source, particularly using yum to update. I'm trying to actually install OpenSips via that method. Does anyone have/know of where I can find an rpm aimed at EL4? And how I can nab libxml2, with the dependencies. I can get an RPM from for libxml2 from ftp://xmlsoft.org/libxml2/, but the dependencies for it are causing me headaches. Any suggestions would be helpful. Thanks. Cheers, Darrin Henshaw This email and its attachments may be confidential and are intended solely for the use of the individual or parties' to whom it is addressed. All comments are solely those of the author and do not necessarily represent those of Ignition. If you are not the intended recipient of this email and its attachments, you must take no action based upon them, nor must you copy or show them to anyone. Please contact the sender if you believe you have received this email in error. Thanks for considering the environmental impact before printing this email. ___ -- 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] T1 problem (call using a .call file)
I still find it weird as even if it is a switch timing problem. Because when is it calling my phone *all the time *and that other area code it *never *calls it. Does that mean asterisk always complete my number in a certain time frame, and the other number no? Also I get the progress code 127 exactly after i move my call file to the outgoing folder, there is no delay, I get it tthe same time I move the move. And also why the call goes through when I put SIP/whatever in the callerid? Does that mean asterisk get to complete the call in the time frame? On Thu, Mar 19, 2009 at 5:54 PM, Danny Nicholas da...@debsinc.com wrote: You can also do a set variable in the call file. I don’t really know how to do that, but you can probably find the command and syntax on voip-info.org. The reason it works on certain numbers has to do with switch timing. If * can complete the call within a certain time frame, all is well. If not, the 127 thing will bite you. You would think we were past that type of thing, but I suppose not. Another thing you might try is changing the 60 to 90 or so on your original call file. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Pascal Bruno *Sent:* Thursday, March 19, 2009 4:42 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] T1 problem (call using a .call file) I dont want to change it within my extensions.conf, because I have many dids, and change them on the fly according to the call i am making. I have a web interface where I fill a form that gets the number I am calling, the caller id and context to go etc... I dont want to keep editing extensions.conf and reload, I want to do it directly in the call file. What I dont understand is WHY it works on certain numbers and not all. That is a problem, it is not normal. On Thu, Mar 19, 2009 at 5:24 PM, Danny Nicholas da...@debsinc.com wrote: GLOBAL_OUTBOUNDCID = XX in extensions.conf [globals] should do the trick -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Thursday, March 19, 2009 3:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] T1 problem (call using a .call file) Pascal Bruno wrote: Also very strange, when in my call file I change the callerid line to SIP/whatever like Danny said, the call go through, but I dont want that, because when I do so, it is displaying the main number on my T1 account as caller id and I dont want that, I want to display one of my other DID as callerid. Then change your caller-id within your dialplan, not the callfile. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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 ___ -- 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] Special Information Tones
GrandCentral/Google Voice does just this, although I have no idea what they use for a back end to make it happen. When someone calls your GC/GV number, it forwards out to a list of numbers you have given the service. You can choose to answer the call, send it on to voicemail, or a couple of other things by hitting 1-5 after you answer. Cary Fitch wrote: From a cell user level perspective... The cell companies are doing it like they think makes sense. If they know your cell is off/out of range they route instantly to VM. They could give 4-10 rings of fake effort, but why. With follow me roaming and such, they want to process the call as fast as possible. If they don't know if the cell is available, they may go through about 4 rings of searching, but beyond that it is time to send it to VM, charge for the call :-), and move on. Ideally, a find me call forwarding system should have a real person identifier and local voice mail. Real person means that all called external numbers should not be assumed to be answered until they send back a DTMF tone. Something like a Background announcement with some silence, waiting for DTMF. It could be a Boing or You have a forwarded call, press any key to accept the call Then the call should be cut through to that extension. Cary Fitch ___ -- 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] ATA recommendation??
Hello, I want to ask that if thee are some ATA decives that i can use to connect mutliple analog phone lines to my VOIP system.. I mean for example an ATA device with 24 ports with 24 independent SIP accounts. For example for some dormitories in my area, i want to put an ATA device and move existing lines to VOIP accounts. Only problem is, if i dont give seperate SIP accounts for all ports, i can not control who is calling where... And the billing system will also be a problem in that case. Tnx... ___ -- 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] T1 problem (call using a .call file)
Don't really know the answer, but these are givens: 1. your phone is (most likely) in the same area code as the asterisk installation 2. NY is most likely not in the same area code. 3. Even though the T1 is a dedicated digital service, the code that handles all of this is/was written to process calls from analog sources for backwards compatibility and therefore would have the timing issue handlers in place even though they don't apply. My research revealed that you might use an exception to stop this, but I didn't really find a good example. You could check viop-info.org or whirlpool to see what they say. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pascal Bruno Sent: Friday, March 20, 2009 9:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] T1 problem (call using a .call file) I still find it weird as even if it is a switch timing problem. Because when is it calling my phone all the time and that other area code it never calls it. Does that mean asterisk always complete my number in a certain time frame, and the other number no? Also I get the progress code 127 exactly after i move my call file to the outgoing folder, there is no delay, I get it tthe same time I move the move. And also why the call goes through when I put SIP/whatever in the callerid? Does that mean asterisk get to complete the call in the time frame? On Thu, Mar 19, 2009 at 5:54 PM, Danny Nicholas da...@debsinc.com wrote: You can also do a set variable in the call file. I don't really know how to do that, but you can probably find the command and syntax on voip-info.org http://voip-info.org/ . The reason it works on certain numbers has to do with switch timing. If * can complete the call within a certain time frame, all is well. If not, the 127 thing will bite you. You would think we were past that type of thing, but I suppose not. Another thing you might try is changing the 60 to 90 or so on your original call file. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pascal Bruno Sent: Thursday, March 19, 2009 4:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] T1 problem (call using a .call file) I dont want to change it within my extensions.conf, because I have many dids, and change them on the fly according to the call i am making. I have a web interface where I fill a form that gets the number I am calling, the caller id and context to go etc... I dont want to keep editing extensions.conf and reload, I want to do it directly in the call file. What I dont understand is WHY it works on certain numbers and not all. That is a problem, it is not normal. On Thu, Mar 19, 2009 at 5:24 PM, Danny Nicholas da...@debsinc.com wrote: GLOBAL_OUTBOUNDCID = XX in extensions.conf [globals] should do the trick -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Thursday, March 19, 2009 3:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] T1 problem (call using a .call file) Pascal Bruno wrote: Also very strange, when in my call file I change the callerid line to SIP/whatever like Danny said, the call go through, but I dont want that, because when I do so, it is displaying the main number on my T1 account as caller id and I dont want that, I want to display one of my other DID as callerid. Then change your caller-id within your dialplan, not the callfile. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com 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 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 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] ATA recommendation??
Grandstream makes an 8-port unit which we've had success with, you could use three of them. Hello, I want to ask that if thee are some ATA decives that i can use to connect mutliple analog phone lines to my VOIP system.. I mean for example an ATA device with 24 ports with 24 independent SIP accounts. For example for some dormitories in my area, i want to put an ATA device and move existing lines to VOIP accounts. Only problem is, if i dont give seperate SIP accounts for all ports, i can not control who is calling where... And the billing system will also be a problem in that case. Tnx... ___ -- 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] ATA recommendation??
Hello, I want to ask that if thee are some ATA decives that i can use to connect mutliple analog phone lines to my VOIP system.. I mean for example an ATA device with 24 ports with 24 independent SIP accounts. For example for some dormitories in my area, i want to put an ATA device and move existing lines to VOIP accounts. Only problem is, if i dont give seperate SIP accounts for all ports, i can not control who is calling where... And the billing system will also be a problem in that case. These are called SIP Gateways. There are several manufacturers who make them. I would suggest Audiocodes, Vega, or Carrier Access as starting points. Yes they come in 24 and 48 port versions. ___ -- 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] OpenSIPS on CentOS
Hello Darrin, Maybe you should ask this question on OpenSIPs mailing list. I have build a rpm for CentOS 5.2 using and updated opensips.spec from svn 1) retrieve opensips.init and opensips.spec-4.4 from https://opensips.svn.sourceforge.net/svnroot/opensips/branches/1.4/packaging/rpm 2) retrieve http://www.opensips.org/pub/opensips/1.4.4/src/opensips-1.4.4-tls_src.tar.gz 3) put opensips.init and opensips-1.4.4-tls_src.tar.gz in /usr/src/redhat/SOURCES 4) put opensips.spec-4.4 in /usr/src/redhat/SPECS 5) run rpmbuild -bb opensips.spec-4.4 (and install missing build dependencies if necessary) ++ Le Friday 20 March 2009 15.19:07 Darrin Henshaw, vous avez écrit : I’ve been looking into OpenSIPS to see if it’s a worthwhile addition to our setup. We’re currently running a cluster, using Heartbeat, between two servers. It works well but I’m interested in seeing if we can improve it. My manager heavily uses RPM’s for installations rather than source, particularly using yum to update. I’m trying to actually install OpenSips via that method. Does anyone have/know of where I can find an rpm aimed at EL4? And how I can nab libxml2, with the dependencies. I can get an RPM from for libxml2 from ftp://xmlsoft.org/libxml2/, but the dependencies for it are causing me headaches. Any suggestions would be helpful. Thanks. -- -- -- Marc LEURENT Ingénieur VoIP DECKPOINT SA Une société du groupe VTX Telecom Rue Eugène-Marziano 15 - 1227 Les Acacias http://www.vtx.ch - marc.leur...@vtx-telecom.ch VTX, votre partenaire telecom proche de vous ! ___ -- 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] Area code 757 Car warranty calls
On Friday 20 March 2009 00:38:45 Cary Fitch wrote: Sure if you can get up stream carriers to cooperate. Just follow the CDRs. But short of a subpoena... or enlightened self interest, like the calls take down a tandem.. (not likely). We could loop the calls back to get ATT's attention, but they would just complain about the loop, not trace them back to the source. Nothing official, but if these are the same clowns who called me earlier this month (and who I filed a complaint on at the DNC registry), then changing their area code may have been a ploy to avoid more complaints. Here is some relevant information on that number: http://whocalled.us/lookup/7025200085 -- 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
[asterisk-users] anyone connection to eoncc
I am trying to connect asterisk 1.4.23 to a customer that has eon from eoncc.com Has anyone done this before? They dont have a sip trunk so we are using sip a SIP extension. All I get is Registration timed out, trying again my register line is something like: register = 5...@ipaddress [VOIP] type=friend dtmfmode=rfc2833 disallow=all allow=ulaw allow=alaw context=smvoice-incoming host=IPADDRESS canreinvite=yes qualify=yes I also tried 5...@ipaddress/5492 I always get the registration timed out. Any thoughts or anyone tried to connect to eon before? Jerry ___ -- 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] Special Information Tones
The message I play to people (or machines) answering is something like DEX Yellow Pages call for Jones Bail Bonds, press [1] This information, along with the caller's phone number, etc., is logged for follow-up. The key, as Cary points out, is to look for DTMF to confirm that the call is not being transferred to voice mail or an unexpected intercept. (In 40 years of doing this stuff, this is the most reliable approach I've found to human/whatever answer-detect.) To further protect from the six-year-old answering the phone or the unfortunate forwarded-to-the-ex-girlfriend's-number, replace [1] with your password. My experience suggests that saying press [1] (or enter your password) results in a little quicker reaction than press any key. Then we remind them to please repeat your greeting. --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cary Fitch Sent: Friday, March 20, 2009 8:25 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Special Information Tones From a cell user level perspective... The cell companies are doing it like they think makes sense. If they know your cell is off/out of range they route instantly to VM. They could give 4-10 rings of fake effort, but why. With follow me roaming and such, they want to process the call as fast as possible. If they don't know if the cell is available, they may go through about 4 rings of searching, but beyond that it is time to send it to VM, charge for the call :-), and move on. Ideally, a find me call forwarding system should have a real person identifier and local voice mail. Real person means that all called external numbers should not be assumed to be answered until they send back a DTMF tone. Something like a Background announcement with some silence, waiting for DTMF. It could be a Boing or You have a forwarded call, press any key to accept the call Then the call should be cut through to that extension. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of drew einhorn Sent: Friday, March 20, 2009 8:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Special Information Tones On Fri, Mar 20, 2009 at 1:53 AM, Gordon Henderson gordon+aster...@drogon.net wrote: On Fri, 20 Mar 2009, Stephen Davies wrote: Hi, Are you sure that Verizon amswers the call? They should play that message as 'early media' without answering, after which they cam clear the call with an appropriate cause code. Yes, They are answering the call, sometimes on the first ring, and taking it away from the lines that should be answering the call. Similar issue in the UK and yes, the carriers do answer the call - because from that second onward thy are taking revenue. BT offer a free voicemailbox on landlines too - for the same reason. So, they really want to answer the phone so they can charge for the call. If we can get them to put one of the Special Information Tones in front of the call, can we make asterisk ignore that false answer and allow the other lines to continue simultaneously ringing until we get a real answer, or it goes to voicemail? Gordon That would work for you and still give callers the audible ,essage they want. Steve On 3/20/09, drew einhorn drew.einh...@gmail.com wrote: I'm having a problem with Verizon Wireless. I would be extremely surprised if I was the only one having this problem. It seems to me that Verizon Wireless might be able to use one of the Special Information Tones to allow us to solve the problem. But I really do not whether my suggestion is compliant with the ITU-T standards. Perhaps someone can give me an expert opinion on whether I should try to get Verizon to implement my suggestion. First I'll describe the problem. I'm trying to implement Single Number Reach. For example, when a call comes in to one of my DIDs, it simultaneously rings on a couple extensions in my home office and a couple of Verizon Wireless cell phone numbers. Everything works just the way it is supposed to if the cell phones are powered up, and within the range of a cell tower. The problem is if a cellphone is turned off, or out of range and unable to talk to a cell tower, Verizon is unable to find the cellphone on their network, Verizon answers the call and plays a recorded message, instead of allowing the number to continue ringing, and allowing one of the voip extensions, or another cellphone to answer the call. Verizon really wants to get rid of the call as quickly as possible to free up their equipment to handle other calls. Unfortunately we spend a lot of time in rural areas where there is no cell tower to talk to. In that case we
Re: [asterisk-users] Area code 757 Car warranty calls
This information appears to be relevant, but useless? --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Friday, March 20, 2009 10:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Area code 757 Car warranty calls On Friday 20 March 2009 00:38:45 Cary Fitch wrote: Sure if you can get up stream carriers to cooperate. Just follow the CDRs. But short of a subpoena... or enlightened self interest, like the calls take down a tandem.. (not likely). We could loop the calls back to get ATT's attention, but they would just complain about the loop, not trace them back to the source. Nothing official, but if these are the same clowns who called me earlier this month (and who I filed a complaint on at the DNC registry), then changing their area code may have been a ploy to avoid more complaints. Here is some relevant information on that number: http://whocalled.us/lookup/7025200085 -- 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] Area code 757 Car warranty calls
Well it will get me off my rant in this forum. Isn't that worth something? Seriously, as users some of us have one 2 line system and others are running multiple systems, absorbing hundreds of thousands of calls a day. Where the %#! warranty calls are coming from or not coming from is useful info. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly Sent: Friday, March 20, 2009 11:23 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Area code 757 Car warranty calls This information appears to be relevant, but useless? --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Friday, March 20, 2009 10:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Area code 757 Car warranty calls On Friday 20 March 2009 00:38:45 Cary Fitch wrote: Sure if you can get up stream carriers to cooperate. Just follow the CDRs. But short of a subpoena... or enlightened self interest, like the calls take down a tandem.. (not likely). We could loop the calls back to get ATT's attention, but they would just complain about the loop, not trace them back to the source. Nothing official, but if these are the same clowns who called me earlier this month (and who I filed a complaint on at the DNC registry), then changing their area code may have been a ploy to avoid more complaints. Here is some relevant information on that number: http://whocalled.us/lookup/7025200085 -- 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 ___ -- 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] Area code 757 Car warranty calls
Cary Fitch wrote: The problem has two prongs - first we are in control of our own landlines and can use asterisk to screen whatever crap we wish before disturbing a real user or allowing a vm to get stored (but it would be nice not to have to). The other issue is we are not for the most part in any kind of control situation of our cellphones, and there is no way to stop that ring from happening and once it does it either needs to be answered or a vm dealt with. This is where the bigger players need to start living up to their responsibilities and not just ignore the problem. Well it will get me off my rant in this forum. Isn't that worth something? Seriously, as users some of us have one 2 line system and others are running multiple systems, absorbing hundreds of thousands of calls a day. Where the %#! warranty calls are coming from or not coming from is useful info. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly Sent: Friday, March 20, 2009 11:23 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Area code 757 Car warranty calls This information appears to be relevant, but useless? --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Friday, March 20, 2009 10:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Area code 757 Car warranty calls On Friday 20 March 2009 00:38:45 Cary Fitch wrote: Sure if you can get up stream carriers to cooperate. Just follow the CDRs. But short of a subpoena... or enlightened self interest, like the calls take down a tandem.. (not likely). We could loop the calls back to get ATT's attention, but they would just complain about the loop, not trace them back to the source. Nothing official, but if these are the same clowns who called me earlier this month (and who I filed a complaint on at the DNC registry), then changing their area code may have been a ploy to avoid more complaints. Here is some relevant information on that number: http://whocalled.us/lookup/7025200085 ___ -- 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] chan_ss7 with ringing, but no voice stream.
Cary Fitch wrote:
SS7 doesn’t send any voice. It sends call info, and tells the switches
which trunk to use for the voice. Trunks are two-way as far as audio
content, though they maybe designated is inbound or outbound trunks.
An audio problem is possibly a NAT or other issue.
Since you are modifying the SS7 code, there could be some error in setting
up the call, but normally the IMT trunks are two way. (Of course they are 4
wire circuits so are two one way paths, but they are matched pairs so,
for practical purposes they would be 1 entity for call set up purposes.)
Actually, the implementations of SS7 support in Asterisk (libss7, and
also the out of tree chan_ss7) include support for signaling and bearer
channels, which is why he's mentioning voice support.
Right now, both implementations function basically like the ISDN code
works - i.e. you have to terminate signaling and bearer channels on the
same box.
Matthew Fredrickson (the libss7 guy :-) )
Digium, Inc.
Cary Fitch
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of lizhong zhu
Sent: Friday, March 20, 2009 2:05 AM
To: asterisk-ss7
Subject: [asterisk-users] chan_ss7 with ringing, but no voice stream.
hello, all of users:
sorry, resend it again for clarifying the message. I have implemented
cha_ss7 in china. initially, the
chan_ss7 can not support the call group. i modify the code.
now the problem is that, both sides can hear the ring, but i
can not hear the voice from each other.
i think the ss7 does not send the voice steam to the destination.
in chan_ss7, i added:
===
static struct ss7_chan *cic_hunt_even_mru(struct linkset*
linkset) {
struct ss7_chan *cur, *prev, *best, *best_prev;
best = NULL;
best_prev = NULL;
for(cur = linkset-idle_list, prev = NULL; cur !=
NULL; prev = cur, cur = cur-next_idle) {
/* Don't select lines that are resetting or
blocked. */
if(!cur-reset_done || (cur-blocked
(BL_LH|BL_RM|BL_RH|BL_UNEQUIPPED|BL_LINKDOWN))) {
continue;
}
/* if((cur-cic % 2) == 0) { */
/*change to this*/
if(((cur-cic % 2) ==
0)0==strcasecmp(cur-link-name,linkname))
{
/* Choose the first idle even circuit,
if any. */
/*end of change*/
best = cur;
best_prev = prev;
break;
} else if(best == NULL) {
/* Remember the first odd circuit, in
case no even circuits are
available. */
best = cur;
best_prev = prev;
}
}
cic_hunt_even_mru if(((cur-cic % 2) ==
0)0==strcasecmp(cur-link-name,linkname))
{
my environment is:
asterisk-1.4.20
chan_ss7-1.0.91
Openvox D410P
===
anyone has an idea for the problem?
please give me some hints!
thanks!
james.zhu
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Re: [asterisk-users] [OpenSIPS-Users] OpenSIPS on CentOS
Hi Darrin, Hi Marc, Darrin, with an OpenSIPS frontend you can do more things actually: 1) move the HA in OpenSIPS - it will be able to re-route if one of the Asterisk boxs is down 2) do LB - you can use in parallel multiple Asterisk boxes and to balance the traffic between 3) you can terminate TLS (from client) and convert to UDP to deliver to Asterisk. Marc, Darrin has a point here - if you want to give a quick try to something, it is nice to be able to install it easily. We already have an APT (for debian) repo up and running (still beta). We could do the same for RPMs or, in the worst case, to generate the packages for download. Also, there are some RPMs (for suse) - see http://www.opensips.org/index.php?n=Resources.Downloads Regards, Bogdan Marc Leurent wrote: Hello Darrin, Maybe you should ask this question on OpenSIPs mailing list. I have build a rpm for CentOS 5.2 using and updated opensips.spec from svn 1) retrieve opensips.init and opensips.spec-4.4 from https://opensips.svn.sourceforge.net/svnroot/opensips/branches/1.4/packaging/rpm 2) retrieve http://www.opensips.org/pub/opensips/1.4.4/src/opensips-1.4.4-tls_src.tar.gz 3) put opensips.init and opensips-1.4.4-tls_src.tar.gz in /usr/src/redhat/SOURCES 4) put opensips.spec-4.4 in /usr/src/redhat/SPECS 5) run rpmbuild -bb opensips.spec-4.4 (and install missing build dependencies if necessary) ++ Le Friday 20 March 2009 15.19:07 Darrin Henshaw, vous avez écrit : I’ve been looking into OpenSIPS to see if it’s a worthwhile addition to our setup. We’re currently running a cluster, using Heartbeat, between two servers. It works well but I’m interested in seeing if we can improve it. My manager heavily uses RPM’s for installations rather than source, particularly using yum to update. I’m trying to actually install OpenSips via that method. Does anyone have/know of where I can find an rpm aimed at EL4? And how I can nab libxml2, with the dependencies. I can get an RPM from for libxml2 from ftp://xmlsoft.org/libxml2/, but the dependencies for it are causing me headaches. Any suggestions would be helpful. 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 Management Application for windows
hello, please any asterisk Management application that use the WXWidget Graphicalll User Interface (GUI) ? the FreePBX is not fully accessible to my screen reader. 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] T1 problem (call using a .call file)
Thanks for your help Don’t really know the answer, but these are “givens”: 1. your phone is (most likely) in the same area code as the asterisk installation My phone has a different area code than the asterisk installation. The asterisk box is in FL and I can call a number in MN but not the 201 or many others 1. 2. NY is most likely not in the same area code. I agree but I could call a MN cell phone for example which works all the time 1. 2. Even though the T1 is a dedicated digital service, the code that handles all of this is/was written to process calls from analog sources for backwards compatibility and therefore would have the timing issue handlers in place even though they don’t apply. My research revealed that you might use an exception to stop this, but I didn’t really find a good example. You could check viop-info.org or whirlpool to see what they say. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Pascal Bruno *Sent:* Friday, March 20, 2009 9:39 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] T1 problem (call using a .call file) I still find it weird as even if it is a switch timing problem. Because when is it calling my phone *all the time *and that other area code it *never *calls it. Does that mean asterisk always complete my number in a certain time frame, and the other number no? Also I get the progress code 127 exactly after i move my call file to the outgoing folder, there is no delay, I get it tthe same time I move the move. And also why the call goes through when I put SIP/whatever in the callerid? Does that mean asterisk get to complete the call in the time frame? On Thu, Mar 19, 2009 at 5:54 PM, Danny Nicholas da...@debsinc.com wrote: You can also do a set variable in the call file. I don’t really know how to do that, but you can probably find the command and syntax on voip-info.org. The reason it works on certain numbers has to do with switch timing. If * can complete the call within a certain time frame, all is well. If not, the 127 thing will bite you. You would think we were past that type of thing, but I suppose not. Another thing you might try is changing the 60 to 90 or so on your original call file. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Pascal Bruno *Sent:* Thursday, March 19, 2009 4:42 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] T1 problem (call using a .call file) I dont want to change it within my extensions.conf, because I have many dids, and change them on the fly according to the call i am making. I have a web interface where I fill a form that gets the number I am calling, the caller id and context to go etc... I dont want to keep editing extensions.conf and reload, I want to do it directly in the call file. What I dont understand is WHY it works on certain numbers and not all. That is a problem, it is not normal. On Thu, Mar 19, 2009 at 5:24 PM, Danny Nicholas da...@debsinc.com wrote: GLOBAL_OUTBOUNDCID = XX in extensions.conf [globals] should do the trick -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Thursday, March 19, 2009 3:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] T1 problem (call using a .call file) Pascal Bruno wrote: Also very strange, when in my call file I change the callerid line to SIP/whatever like Danny said, the call go through, but I dont want that, because when I do so, it is displaying the main number on my T1 account as caller id and I dont want that, I want to display one of my other DID as callerid. Then change your caller-id within your dialplan, not the callfile. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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:
Re: [asterisk-users] Area code 757 Car warranty calls
Verizon wireless filed a lawsuit against the perpetrators of the car warranty scam. I hope to hell they win. http://www.foxnews.com/story/0,2933,501404,00.html Cary Fitch wrote: The problem has two prongs - first we are in control of our own landlines and can use asterisk to screen whatever crap we wish before disturbing a real user or allowing a vm to get stored (but it would be nice not to have to). The other issue is we are not for the most part in any kind of control situation of our cellphones, and there is no way to stop that ring from happening and once it does it either needs to be answered or a vm dealt with. This is where the bigger players need to start living up to their responsibilities and not just ignore the problem. Well it will get me off my rant in this forum. Isn't that worth something? Seriously, as users some of us have one 2 line system and others are running multiple systems, absorbing hundreds of thousands of calls a day. Where the %#! warranty calls are coming from or not coming from is useful info. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly Sent: Friday, March 20, 2009 11:23 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Area code 757 Car warranty calls This information appears to be relevant, but useless? --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Friday, March 20, 2009 10:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Area code 757 Car warranty calls On Friday 20 March 2009 00:38:45 Cary Fitch wrote: Sure if you can get up stream carriers to cooperate. Just follow the CDRs. But short of a subpoena... or enlightened self interest, like the calls take down a tandem.. (not likely). We could loop the calls back to get ATT's attention, but they would just complain about the loop, not trace them back to the source. Nothing official, but if these are the same clowns who called me earlier this month (and who I filed a complaint on at the DNC registry), then changing their area code may have been a ploy to avoid more complaints. Here is some relevant information on that number: http://whocalled.us/lookup/7025200085 ___ -- 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] Queues Announce help request.
I am trying to get a queue to do more than just play music and hold calls. Specifically, making some comforting voice announcements would be nice. Below is the queues.conf file relevant portions. Member phone number is munged to protect the guilty. We shouldn't need the announcement source info, but I have been trying everything. The problem is with the member busy, we get no voice announcements. (For test purposes is being on hold busy? We have also just laid the phone on the desk.) We will settle for expected hold time, Thank you announcements, Position in queue, or Dow Jones 30 Industrials news. :-) Anyone have a tip? Cary Fitch Here are some relevant errors from console. [Mar 20 12:16:16] WARNING[4317]: chan_sip.c:12406 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog '495f59f65dec70c67849cf4f0cd09...@72.49.176.4'. Giving up. -- SIP/3617001000-009a3930 is circuit-busy -- Nobody picked up in 6 ms -- Exiting on time-out cycle - [Mar 20 12:16:16] WARNING[4317]: chan_sip.c:12949 handle_response: Remote host can't match request CANCEL to call '495f59f65dec70c67849cf4f0cd09...@72.49.176.4'. Giving up. -- Stopped music on hold on SIP/3617001401-fc0359f0 [usa-queue] queue-youarenext = queue-youarenext queue-thereare = there-thereare queue-callswaiting = queue-callswaiting queue-holdtime = queue-holdtime queue-thankyou = queue-thankyou music = default maxlen = 0 strategy = ringall context = leave-message periodic-announce = thank-you-message periodic-announce-frequency = 30 announce-frequency = 30 announce-holdtime = yes announce-round-seconds = 10 joinempty = no leavewhenempty = yes retry = 30 timeout = 300 member = SIP/3617001402 ___ -- 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] Area code 757 Car warranty calls
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Moffett Sent: Friday, March 20, 2009 12:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Area code 757 Car warranty calls Verizon wireless filed a lawsuit against the perpetrators of the car warranty scam. I hope to hell they win. http://www.foxnews.com/story/0,2933,501404,00.html [Cary Fitch] That rings a bell, but I wonder if the afore mentioned So. Cal. Individuals are involved. Verizon's suit doesn't seem to have lowered the volume of calls much. Perhaps there are multiple perps. CF ___ -- 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] Area code 757 Car warranty calls
On Fri, Mar 20, 2009 at 1:23 PM, Adam Moffett a...@plexicomm.net wrote: Verizon wireless filed a lawsuit against the perpetrators of the car warranty scam. I hope to hell they win. http://www.foxnews.com/story/0,2933,501404,00.html Thanks for the link, now we got a name to the scam fwitw :P Cary Fitch wrote: The problem has two prongs - first we are in control of our own landlines and can use asterisk to screen whatever crap we wish before disturbing a real user or allowing a vm to get stored (but it would be nice not to have to). The other issue is we are not for the most part in any kind of control situation of our cellphones, and there is no way to stop that ring from happening and once it does it either needs to be answered or a vm dealt with. This is where the bigger players need to start living up to their responsibilities and not just ignore the problem. Well it will get me off my rant in this forum. Isn't that worth something? Seriously, as users some of us have one 2 line system and others are running multiple systems, absorbing hundreds of thousands of calls a day. Where the %#! warranty calls are coming from or not coming from is useful info. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly Sent: Friday, March 20, 2009 11:23 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Area code 757 Car warranty calls This information appears to be relevant, but useless? --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Friday, March 20, 2009 10:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Area code 757 Car warranty calls On Friday 20 March 2009 00:38:45 Cary Fitch wrote: Sure if you can get up stream carriers to cooperate. Just follow the CDRs. But short of a subpoena... or enlightened self interest, like the calls take down a tandem.. (not likely). We could loop the calls back to get ATT's attention, but they would just complain about the loop, not trace them back to the source. Nothing official, but if these are the same clowns who called me earlier this month (and who I filed a complaint on at the DNC registry), then changing their area code may have been a ploy to avoid more complaints. Here is some relevant information on that number: http://whocalled.us/lookup/7025200085 ___ -- 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
Re: [asterisk-users] [OpenSIPS-Users] OpenSIPS on CentOS
Hello Bogdan, I have set a small rpm repository for opensips 4.4 for CentOS (with el5) x86_64 (and later i386 32bits) Simply visit http://centos.leurent.eu/ and read the README.txt Maybe we could just do the same on the official OpenSIPs website? ++ Le Friday 20 March 2009 18.12:25 Bogdan-Andrei Iancu, vous avez écrit : Hi Darrin, Hi Marc, Darrin, with an OpenSIPS frontend you can do more things actually: 1) move the HA in OpenSIPS - it will be able to re-route if one of the Asterisk boxs is down 2) do LB - you can use in parallel multiple Asterisk boxes and to balance the traffic between 3) you can terminate TLS (from client) and convert to UDP to deliver to Asterisk. Marc, Darrin has a point here - if you want to give a quick try to something, it is nice to be able to install it easily. We already have an APT (for debian) repo up and running (still beta). We could do the same for RPMs or, in the worst case, to generate the packages for download. Also, there are some RPMs (for suse) - see http://www.opensips.org/index.php?n=Resources.Downloads Regards, Bogdan Marc Leurent wrote: Hello Darrin, Maybe you should ask this question on OpenSIPs mailing list. I have build a rpm for CentOS 5.2 using and updated opensips.spec from svn 1) retrieve opensips.init and opensips.spec-4.4 from https://opensips.svn.sourceforge.net/svnroot/opensips/branches/1.4/packaging/rpm 2) retrieve http://www.opensips.org/pub/opensips/1.4.4/src/opensips-1.4.4-tls_src.tar.gz 3) put opensips.init and opensips-1.4.4-tls_src.tar.gz in /usr/src/redhat/SOURCES 4) put opensips.spec-4.4 in /usr/src/redhat/SPECS 5) run rpmbuild -bb opensips.spec-4.4 (and install missing build dependencies if necessary) ++ Le Friday 20 March 2009 15.19:07 Darrin Henshaw, vous avez écrit : I’ve been looking into OpenSIPS to see if it’s a worthwhile addition to our setup. We’re currently running a cluster, using Heartbeat, between two servers. It works well but I’m interested in seeing if we can improve it. My manager heavily uses RPM’s for installations rather than source, particularly using yum to update. I’m trying to actually install OpenSips via that method. Does anyone have/know of where I can find an rpm aimed at EL4? And how I can nab libxml2, with the dependencies. I can get an RPM from for libxml2 from ftp://xmlsoft.org/libxml2/, but the dependencies for it are causing me headaches. Any suggestions would be helpful. Thanks. -- -- -- Marc LEURENT Ingénieur VoIP DECKPOINT SA Une société du groupe VTX Telecom Rue Eugène-Marziano 15 - 1227 Les Acacias http://www.vtx.ch - marc.leur...@vtx-telecom.ch VTX, votre partenaire telecom proche de vous ! ___ -- 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] Queues Announce help request.
The most popular answer I've seen here is to replace the regular music with a streamed audio feed which can be anything you have access to. I'd try and give you details, but they wouldn't be correct. This information is pretty easy to locate in the digium site, viop-info.org or google. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cary Fitch Sent: Friday, March 20, 2009 12:32 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Queues Announce help request. I am trying to get a queue to do more than just play music and hold calls. Specifically, making some comforting voice announcements would be nice. Below is the queues.conf file relevant portions. Member phone number is munged to protect the guilty. We shouldn't need the announcement source info, but I have been trying everything. The problem is with the member busy, we get no voice announcements. (For test purposes is being on hold busy? We have also just laid the phone on the desk.) We will settle for expected hold time, Thank you announcements, Position in queue, or Dow Jones 30 Industrials news. :-) Anyone have a tip? Cary Fitch Here are some relevant errors from console. [Mar 20 12:16:16] WARNING[4317]: chan_sip.c:12406 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog '495f59f65dec70c67849cf4f0cd09...@72.49.176.4'. Giving up. -- SIP/3617001000-009a3930 is circuit-busy -- Nobody picked up in 6 ms -- Exiting on time-out cycle - [Mar 20 12:16:16] WARNING[4317]: chan_sip.c:12949 handle_response: Remote host can't match request CANCEL to call '495f59f65dec70c67849cf4f0cd09...@72.49.176.4'. Giving up. -- Stopped music on hold on SIP/3617001401-fc0359f0 [usa-queue] queue-youarenext = queue-youarenext queue-thereare = there-thereare queue-callswaiting = queue-callswaiting queue-holdtime = queue-holdtime queue-thankyou = queue-thankyou music = default maxlen = 0 strategy = ringall context = leave-message periodic-announce = thank-you-message periodic-announce-frequency = 30 announce-frequency = 30 announce-holdtime = yes announce-round-seconds = 10 joinempty = no leavewhenempty = yes retry = 30 timeout = 300 member = SIP/3617001402 ___ -- 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] Queues Announce help request.
- Cary Fitch ca...@usawide.net wrote: I am trying to get a queue to do more than just play music and hold calls. Specifically, making some comforting voice announcements would be nice. You may want to take the quotes off of the filenames in your queues.conf config file... they're not needed, and could very well be causing your problems. -- Jared Smith Training Manager Digium, Inc. ___ -- 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] Looking for clues to this error message
[Mar 20 12:45:33] WARNING[4940]: app_queue.c:3136 try_calling: The device state of this queue member, SIP/3617001000, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. [Cary Fitch] We are running 1.4.22 and this message popped up in console. It could be causing our Queues announcement problem, because if all members don't show busy.. there will be no announcements. However, I see no references to such an issue in any upgrade documents I have found. Any one have a tip? Thanks Cary Fitch ___ -- 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] Queues Announce help request.
On 3/20/09, Cary Fitch ca...@usawide.net wrote: I am trying to get a queue to do more than just play music and hold calls. Specifically, making some comforting voice announcements would be nice. Below is the queues.conf file relevant portions. Member phone number is munged to protect the guilty. We shouldn't need the announcement source info, but I have been trying everything. The problem is with the member busy, we get no voice announcements. (For test purposes is being on hold busy? We have also just laid the phone on the desk.) We will settle for expected hold time, Thank you announcements, Position in queue, or Dow Jones 30 Industrials news. :-) Anyone have a tip? Cary Fitch I just thought I'd mention that ViciDial has the ability to play a periodic announcement on inbound queue calls, as well as music on hold, place in line, estimated hold time and lots of other inbound-only features. ViciDial does not use Asterisk Queues so the way we got it working probably wouldn't help you much, but I just wanted to mention it's functionality and that it is open source if you wanted to give it a try. Thanks, MATT--- ___ -- 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] [OpenSIPS-Users] Asterisk is not designed for University with largeuser base?
Hi Yehavi, Please see my inline comments: Yehavi Bourvine wrote: Hello, Sorry for the delay - was out of office. I also cross-posting it to OpenSIPS list. I have a small pilot (20-30 phones) which also does some sort of SIP to PRI transcode for our old PBX. The pilot is base on Asterisk and mostly Polycom-501 phones. It works quite well, but I have a few minor/missing issues: - I have the RPID patch, and unattended transfers fails with it. - No SLA, only BLF. I know there is SLA, but it is cumbersome to deploy. - Confference is limited to 3 participants. I guess I can do more with external server but didn't manage yet to make it working. - No busy dial again which is required by our users. Now, to the original issue: I tried adding 1000 extensions to the SIP database, and then use SIPP to send one REGISTER for each extension. After doing so Asterisk still worked, but it was continously accessing the database for all these extensions, just polling them. This raised a red flag to me, and I decided to check the following config: OpenSIPS/Kamailo/etc. as registrar and SIP switch for the phones, while using Asterisk only for media related issues (which is the common suggestion here). Actual this is the natural way of doing. You have two pieces of software, with different purposes, but complementary in the same time. Asterisk is an IPPBX handling media and implementing a lot of nice class5 features - and an PBX is not more large numbers of lines OpenSIPS is an softswitch, no media, limited class 5 features, but nice routing and able to handle hundreds of thousands of line and subscribers See: http://www.opensips.org/index.php?n=Resources.DocsTutLoadbalancing Now, I have new problems: - SLA works, but very fragile. - Not BLF, although I think it will be solve with the dialog handling on OpenSIPS 1.5 yes, there is such a module (thanks to the combination of dialog and presence features) - Same confference and busy dial problem. Next week our management is going to decide (I hope...) how to proceed: Do nothing (stay with the Nortel as we are tight on budget), go to open source or to a commercial solution. Although a commercial solution allows me so sleep well at night, I am going to recommend the open source direction. If accepted, then I will continue the development and you'll hear me quite a lot here asking hard questions :-) Well, a commercial solution does not exclude open source - there are lot of companies offering commercial products based on OSS. So, you can get a good price (no licenses) and you can still sleep well :). James Body from Truphone was asked (during a VON when he had an OpenSER talk) if using OSS is cheaper - the answer was no, it is not, but the difference comes in what you get in the end - instead of paying for 70% licence code and 30% for customization, with OSS you can pay 100% for customization/tunings. So, at the end you get exactly what you want and not a compromise solution (cost versus requirements). BTW, If I didn't say so far: we have around 8,000 extensions on 4 Notel PBX'es, using around 10 PRI's to the world. Regards, Bogdan Regards, __Yehavi: 2009/3/17 Vincent Li vincent.mc.li http://vincent.mc.li/@gmail.com http://gmail.com/ On Tue, 17 Mar 2009, Yehavi Bourvine wrote: Hello' I am at the same situation as you. I also work at a university and we have over 8.000 extensions on a Nortel PBX. I also run a small Asterisk pilot. I am using a realtime users database and the main problem is that Aaterisk does too mcuh database access to inquire for the currently registered users. (I am using direct RTP path between the phones so this is not a limiting issue here). I am checking now a combination of OpenSIPS and Asterisk, where OpenSIPS will serve the phones and Asterisk the more complicate things (voicemail, transcoding, etc.). OpenSIPS still lacks some of Asterisk features, but they are being worked on. Regards, __Yehavi: Hi Yehavi, Could you please keep us informed with your research, That would be very interesting case that all other Universities could study. There seems no known large Asterisk deployment in University enviroment at this time. Regards, ___ Users mailing list us...@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
[asterisk-users] Asterisk 1.6.0.7-rc2, 1.6.1.0-rc3, 1.6.2.0-beta1 Asterisk-Addons 1.6.0.2-rc1, 1.6.1.0-rc3 Now Available
The Asterisk.org development team is pleased to announced the release of Asterisk release candidates 1.6.0.7-rc2, 1.6.1.0-rc3, and beta release 1.6.2.0-beta1. Additionally, new release candidates of Asterisk-Addons 1.6.0.2-rc1 and 1.6.1.0-rc3 have been created. Note that the 1.6.1 series of Asterisk-Addons is compatible with both Asterisk 1.6.1 and 1.6.2 branches. These releases are available for immediate download from http://downloads.digium.com/. These Asterisk releases improve compatibility with T.38 switchovers internally; fixes an issue with Asterisk using poll() on OSX systems when it should not; allows chan_h323 to be built against both OpenH323 and H323Plus libraries (while simplifying the build process); and improve behavior of ast_answer() which was problematic for T.38 re-INVITES and other sorts of channel operations. Additionally, other bugs have also been resolved in these release candidates. The Asterisk-Addons release candidates fix a few minor issues since the last set of release candidates. The first beta release of the Asterisk 1.6.2 series is now available. You can get more information about the new features and various changes in this release at: http://svn.digium.com/view/asterisk/tags/1.6.2.0-beta1/CHANGES?view=co And if you're upgrading from previous versions of Asterisk see this file: http://svn.digium.com/view/asterisk/tags/1.6.2.0-beta1/UPGRADE.txt?view=co ChangeLogs for the various releases candidates and beta are available at: http://svn.digium.com/view/asterisk/tags/1.6.0.7-rc2/ChangeLog?view=co http://svn.digium.com/view/asterisk/tags/1.6.1.0-rc3/ChangeLog?view=co http://svn.digium.com/view/asterisk/tags/1.6.2.0-beta1/ChangeLog?view=co http://svn.digium.com/view/asterisk-addons/tags/1.6.0.2-rc1/ChangeLog?view=co http://svn.digium.com/view/asterisk-addons/tags/1.6.1.0-rc3/ChangeLog?view=co Issues discovered in testing of these release candidates and beta can be reported at http://bugs.digium.com Thank you for your continued support of Asterisk! ___ -- 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] Area code 757 Car warranty calls
Jon Pounder wrote: Cary Fitch wrote: The problem has two prongs - first we are in control of our own landlines and can use asterisk to screen whatever crap we wish before disturbing a real user or allowing a vm to get stored (but it would be nice not to have to). The other issue is we are not for the most part in any kind of control situation of our cellphones, and there is no way to stop that ring from happening and once it does it either needs to be answered or a vm dealt with. This is where the bigger players need to start living up to their responsibilities and not just ignore the problem. Well it will get me off my rant in this forum. Isn't that worth something? Seriously, as users some of us have one 2 line system and others are running multiple systems, absorbing hundreds of thousands of calls a day. Where the %#! warranty calls are coming from or not coming from is useful info. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly Sent: Friday, March 20, 2009 11:23 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Area code 757 Car warranty calls This information appears to be relevant, but useless? --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Friday, March 20, 2009 10:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Area code 757 Car warranty calls On Friday 20 March 2009 00:38:45 Cary Fitch wrote: Sure if you can get up stream carriers to cooperate. Just follow the CDRs. But short of a subpoena... or enlightened self interest, like the calls take down a tandem.. (not likely). We could loop the calls back to get ATT's attention, but they would just complain about the loop, not trace them back to the source. Nothing official, but if these are the same clowns who called me earlier this month (and who I filed a complaint on at the DNC registry), then changing their area code may have been a ploy to avoid more complaints. Here is some relevant information on that number: http://whocalled.us/lookup/7025200085 I realize that finding these (insert foul and derogatory expletive), but if we do maybe a public whipping with a cat-o-nine tails on the six O'Clock news? It is my phone, I pay for the service I should not have to answer (or even filter out) calls from some idiot that has absolutely no business calling me in the first place. There are many other avenues of advertising that are not invasive of my privacy and do not require me to pay for the call in the case of a cell phone number. Maybe sending a bill to some of these jerks for all the cell calls they have made will hit them where it hurts. Just my opinion JohnM ___ -- 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] Music on Hold doesn't play back for external callers
Hey all; I am experiencing an issue with music on Hold. I am running asterisk version 1.4.22, and have a default script set up in two places for MoH playback. For internal devices to my network that are SIP peering with asterisk, they simply dial 123 and hear the MoH music immediately. I'm using the default setup, where it just plays the wave files in the /var/lib/asterisk/moh directory. I also have in my General settings in sip.conf to use the default musicclass. On-net context script for MoH in extensions.conf: exten = 123,1,Answer() exten = 123,2,MusicOnHold(default) Calls from the PSTN are matching this Moh setup in extensions.conf: exten = _1720XXX,1,Answer() exten = _1720XXX,2,MusicOnHold(default) Output captured when internal devices attempt to dial number associated with MoH:(MoH is heard) -- Executing [...@home:1] Answer(SIP/10.1.1.254-b77089b0, ) in new stack -- Executing [...@home:2] MusicOnHold(SIP/10.1.1.254-b77089b0, default) in new stack -- Started music on hold, class 'default', on SIP/10.1.1.254-b77089b0 Output captured when external callers dial into asterisk from the PSTN, they get dead air instead of MoH: -- Executing [1720...@home:1] Answer(SIP/1720XXX-b7704a38, ) in new stack -- Executing [1720...@home:2] MusicOnHold(SIP/1720XXX-b7704a38, default) in new stack -- Started music on hold, class 'default', on SIP/1720XXX-b7704a38 You see rule 1(answer) and rule 2(moh) being hit on both call attempts, but the external caller continues to get dead air. has anyone seen this one before? The internal devices requesting MoH are using the SCCP protocol load image(registered elsewhere) and then they are SIP trunked to Asterisk. Asterisk uses a second separate SIP trunk to peer with the PSTN. -Greg _ Windows Live™ SkyDrive: Get 25 GB of free online storage. http://windowslive.com/online/skydrive?ocid=TXT_TAGLM_WL_skydrive_032009___ -- 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] 1.6.2 beta 1 crash
Hi,
I'm starting testing 1.6.2 beta. CentOs 5.2
I found my first crash, first I have
[Mar 20 20:30:41] WARNING[11201]: res_config_mysql.c:611 update_mysql:
Attempted to update column 'useragent' in table 'sip', but column does not
exist!
[Mar 20 20:30:41] ERROR[11201]: res_config_mysql.c:581 update_mysql: MySQL
RealTime: Updating on column 'lastms', but that column does not exist within
the table 'sip'!
I would like to know if this new fields corresponds to previous ones, so I
can delete deprecated ones and add this new to my tables.
CRASH:
localhost*CLI *** glibc detected *** /usr/sbin/asterisk: realloc(): invalid
next size: 0x08abff28 ***
=== Backtrace: =
/lib/libc.so.6[0xdfa440]
/lib/libc.so.6(realloc+0x1a7)[0xdfb377]
/usr/sbin/asterisk(__ast_str_helper+0x75)[0x81403b5]
/usr/sbin/asterisk(ast_str_append+0x31)[0x814d531]
/usr/lib/asterisk/modules/func_odbc.so[0x14c1ee5]
/usr/sbin/asterisk(ast_func_read+0x100)[0x81009e0]
/usr/sbin/asterisk(pbx_substitute_variables_helper_full+0x29d)[0x81070dd]
/usr/sbin/asterisk[0x810797a]
/usr/sbin/asterisk[0x810a0a4]
/usr/sbin/asterisk[0x810bc20]
/usr/sbin/asterisk[0x814c92b]
/lib/libpthread.so.0[0x4ef46b]
/lib/libc.so.6(clone+0x5e)[0xe5fdbe]
#0 0x00c6c402 in __kernel_vsyscall ()
(gdb) bt
#0 0x00c6c402 in __kernel_vsyscall ()
#1 0x0019ad20 in raise () from /lib/libc.so.6
#2 0x0019c631 in abort () from /lib/libc.so.6
#3 0x001d2e6b in __libc_message () from /lib/libc.so.6
#4 0x001dd440 in _int_realloc () from /lib/libc.so.6
#5 0x001de377 in realloc () from /lib/libc.so.6
#6 0x081403b5 in __ast_str_helper (buf=0xb6f9ed54, max_len=0, append=1,
fmt=0x60f0ac3 ,, ap=0xb6f9ea1c )
at /usr/src/integra/asterisk-1.6.2.0-beta1/include/asterisk/utils.h:493
#7 0x0814d531 in ast_str_append (buf=0xb6f9ed54, max_len=0, fmt=0x60f0ac3
,) at
/usr/src/integra/asterisk-1.6.2.0-beta1/include/asterisk/strings.h:749
#8 0x0605 in acf_odbc_read (chan=0x8a527e8, cmd=0xb6f9ed90
ODBC_MSSQL, s=0xb6f9ed9b IVR_TEST_Asterisk 1001, buf=0xb6f9ee10 1,0,
len=4096)
at func_odbc.c:545
#9 0x081009e0 in ast_func_read (chan=0x8a527e8, function=0xb6f9fe20
ODBC_MSSQL(IVR_TEST_Asterisk 1001), workspace=0xb6f9ee10 1,0, len=4096)
at pbx.c:3346
#10 0x081070dd in pbx_substitute_variables_helper_full (c=0x8a527e8,
headp=0x8a52900, cp1=0x8a0fae8 HASH(Cliente)=${ODBC_MSSQL(IVR_TEST_Asterisk
${ANI})},
cp2=0xb6fa3f02 , count=8177, used=0xb6fa61a8) at pbx.c:3481
#11 0x0810797a in pbx_extension_helper (c=0x8a527e8, con=0x0,
context=0x8a52a58 outgoing, exten=0x8a52aa8 141, priority=11, label=0x0,
callerid=0x8a4b0a0 1001,
action=E_SPAWN, found=0xb6fa6338, combined_find_spawn=1) at pbx.c:3576
#12 0x0810a0a4 in __ast_pbx_run (c=0x8a527e8, args=0x0) at pbx.c:4137
#13 0x0810bc20 in pbx_thread (data=0x8a527e8) at pbx.c:4514
#14 0x0814c92b in dummy_start (data=0x8a53758) at utils.c:968
#15 0x0011d46b in start_thread () from /lib/libpthread.so.0
#16 0x00242dbe in clone () from /lib/libc.so.6
If more info is needed or I should submit a bug report, please let me know.
Regards,
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[asterisk-users] Asterisk with encryption
Dear all, I want to know if anybody has implented an Asterisk server (1.4 or 1.6) with SRTP and SIP with TLS, in order to encrypt both signaling and voice packets. Is it possible ?? And in the affirmative case, does encryption increase the delay and so the voice quality becomes wrong ??? Thanks a lot. Alejandro ___ -- 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] 1.6.2 beta 1 crash
Seems to be a crash with func_odbc.
Let me know what info you need to check it.
Thnks
From: Sebastian [mailto:s...@adinet.com.uy]
Sent: viernes, 20 de marzo de 2009 10:37 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: 1.6.2 beta 1 crash
Hi,
I'm starting testing 1.6.2 beta. CentOs 5.2
I found my first crash, first I have
[Mar 20 20:30:41] WARNING[11201]: res_config_mysql.c:611 update_mysql:
Attempted to update column 'useragent' in table 'sip', but column does not
exist!
[Mar 20 20:30:41] ERROR[11201]: res_config_mysql.c:581 update_mysql: MySQL
RealTime: Updating on column 'lastms', but that column does not exist within
the table 'sip'!
I would like to know if this new fields corresponds to previous ones, so I
can delete deprecated ones and add this new to my tables.
CRASH:
localhost*CLI *** glibc detected *** /usr/sbin/asterisk: realloc(): invalid
next size: 0x08abff28 ***
=== Backtrace: =
/lib/libc.so.6[0xdfa440]
/lib/libc.so.6(realloc+0x1a7)[0xdfb377]
/usr/sbin/asterisk(__ast_str_helper+0x75)[0x81403b5]
/usr/sbin/asterisk(ast_str_append+0x31)[0x814d531]
/usr/lib/asterisk/modules/func_odbc.so[0x14c1ee5]
/usr/sbin/asterisk(ast_func_read+0x100)[0x81009e0]
/usr/sbin/asterisk(pbx_substitute_variables_helper_full+0x29d)[0x81070dd]
/usr/sbin/asterisk[0x810797a]
/usr/sbin/asterisk[0x810a0a4]
/usr/sbin/asterisk[0x810bc20]
/usr/sbin/asterisk[0x814c92b]
/lib/libpthread.so.0[0x4ef46b]
/lib/libc.so.6(clone+0x5e)[0xe5fdbe]
#0 0x00c6c402 in __kernel_vsyscall ()
(gdb) bt
#0 0x00c6c402 in __kernel_vsyscall ()
#1 0x0019ad20 in raise () from /lib/libc.so.6
#2 0x0019c631 in abort () from /lib/libc.so.6
#3 0x001d2e6b in __libc_message () from /lib/libc.so.6
#4 0x001dd440 in _int_realloc () from /lib/libc.so.6
#5 0x001de377 in realloc () from /lib/libc.so.6
#6 0x081403b5 in __ast_str_helper (buf=0xb6f9ed54, max_len=0, append=1,
fmt=0x60f0ac3 ,, ap=0xb6f9ea1c )
at /usr/src/integra/asterisk-1.6.2.0-beta1/include/asterisk/utils.h:493
#7 0x0814d531 in ast_str_append (buf=0xb6f9ed54, max_len=0, fmt=0x60f0ac3
,) at
/usr/src/integra/asterisk-1.6.2.0-beta1/include/asterisk/strings.h:749
#8 0x0605 in acf_odbc_read (chan=0x8a527e8, cmd=0xb6f9ed90
ODBC_MSSQL, s=0xb6f9ed9b IVR_TEST_Asterisk 1001, buf=0xb6f9ee10 1,0,
len=4096)
at func_odbc.c:545
#9 0x081009e0 in ast_func_read (chan=0x8a527e8, function=0xb6f9fe20
ODBC_MSSQL(IVR_TEST_Asterisk 1001), workspace=0xb6f9ee10 1,0, len=4096)
at pbx.c:3346
#10 0x081070dd in pbx_substitute_variables_helper_full (c=0x8a527e8,
headp=0x8a52900, cp1=0x8a0fae8 HASH(Cliente)=${ODBC_MSSQL(IVR_TEST_Asterisk
${ANI})},
cp2=0xb6fa3f02 , count=8177, used=0xb6fa61a8) at pbx.c:3481
#11 0x0810797a in pbx_extension_helper (c=0x8a527e8, con=0x0,
context=0x8a52a58 outgoing, exten=0x8a52aa8 141, priority=11, label=0x0,
callerid=0x8a4b0a0 1001,
action=E_SPAWN, found=0xb6fa6338, combined_find_spawn=1) at pbx.c:3576
#12 0x0810a0a4 in __ast_pbx_run (c=0x8a527e8, args=0x0) at pbx.c:4137
#13 0x0810bc20 in pbx_thread (data=0x8a527e8) at pbx.c:4514
#14 0x0814c92b in dummy_start (data=0x8a53758) at utils.c:968
#15 0x0011d46b in start_thread () from /lib/libpthread.so.0
#16 0x00242dbe in clone () from /lib/libc.so.6
If more info is needed or I should submit a bug report, please let me know.
Regards,
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Re: [asterisk-users] ATA recommendation??
On Mar 20, 2009, at 7:49 AM, Oguzhan Kayhan wrote: Hello, I want to ask that if thee are some ATA decives that i can use to connect mutliple analog phone lines to my VOIP system.. I mean for example an ATA device with 24 ports with 24 independent SIP accounts. For example for some dormitories in my area, i want to put an ATA device and move existing lines to VOIP accounts. Only problem is, if i dont give seperate SIP accounts for all ports, i can not control who is calling where... And the billing system will also be a problem in that case. The Cisco SPA8000 is an 8-port unit with a low per port cost. -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ -- 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] Looking for clues to this error message
Hi Cary, I glanced at the code in main/devicestate.c and it seems the AST_DEVICE_NOT_INUSE is set when the agent channel does not exist ... and that's most likely true when the agent is not logged in or you had asterisk reloaded and the information about agent has been lost ... (true only for callback agents) Are you using callback agents ? Can you describe the Queues announcement problem in more detail ? Martin On Fri, Mar 20, 2009 at 1:05 PM, Cary Fitch ca...@usawide.net wrote: [Mar 20 12:45:33] WARNING[4940]: app_queue.c:3136 try_calling: The device state of this queue member, SIP/3617001000, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. [Cary Fitch] We are running 1.4.22 and this message popped up in console. It could be causing our Queues announcement problem, because if all members don't show busy.. there will be no announcements. However, I see no references to such an issue in any upgrade documents I have found. Any one have a tip? Thanks Cary Fitch ___ -- 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] Music on Hold doesn't play back for external callers
Do you have zaptel timing working ? (or dahdi) ? If no timing is available then if there's no incoming audio frames coming from the external SIP channel then no outgoing audio will be produced (even if you have MOH application working) The trigger to fire the outgoing audio frame comes from the incoming audio frame. So check if the Answer + Echo application work. Also if you have VAD which means your SIP device doesn't send audio frames when there's silence detected = you'd also have the same problem = not see audio going to your phone. Martin On Fri, Mar 20, 2009 at 6:07 PM, Greg Hinson greg_hin...@msn.com wrote: Hey all; I am experiencing an issue with music on Hold. I am running asterisk version 1.4.22, and have a default script set up in two places for MoH playback. For internal devices to my network that are SIP peering with asterisk, they simply dial 123 and hear the MoH music immediately. I'm using the default setup, where it just plays the wave files in the /var/lib/asterisk/moh directory. I also have in my General settings in sip.conf to use the default musicclass. *On-net context script for MoH in extensions.conf:* exten = 123,1,Answer() exten = 123,2,MusicOnHold(default) *Calls from the PSTN are matching this Moh setup in extensions.conf:* exten = _1720XXX,1,Answer() exten = _1720XXX,2,MusicOnHold(default) *Output captured when internal devices attempt to dial number associated with MoH:(MoH is heard)* -- Executing [...@home:1] Answer(SIP/10.1.1.254-b77089b0, ) in new stack -- Executing [...@home:2] MusicOnHold(SIP/10.1.1.254-b77089b0, default) in new stack -- Started music on hold, class 'default', on SIP/10.1.1.254-b77089b0 *Output captured when external callers dial into asterisk from the PSTN, they get dead air instead of MoH:* -- Executing [1720...@home:1] Answer(SIP/1720XXX-b7704a38, ) in new stack -- Executing [1720...@home:2] MusicOnHold(SIP/1720XXX-b7704a38, default) in new stack -- Started music on hold, class 'default', on SIP/1720XXX-b7704a38 You see rule 1(answer) and rule 2(moh) being hit on both call attempts, but the external caller continues to get dead air. has anyone seen this one before? The internal devices requesting MoH are using the SCCP protocol load image(registered elsewhere) and then they are SIP trunked to Asterisk. Asterisk uses a second separate SIP trunk to peer with the PSTN. -Greg -- Windows Live™ SkyDrive: Get 25 GB of free online storage. Check it out.http://windowslive.com/online/skydrive?ocid=TXT_TAGLM_WL_skydrive_032009 ___ -- 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
