Re: [Asterisk-Users] CallerID name lookup AGI script
When I run the script from the command line, I get the following error:
[EMAIL PROTECTED] agi-bin]# ./cid_rewrite.php
Parse error: parse error, expecting `T_OLD_FUNCTION' or
`T_FUNCTION' or `T_VAR' or `'}'' in
/var/lib/asterisk/agi-bin/astlib_jm.php on line 73
Fatal error: Cannot instantiate non-existent class: agi in
/var/lib/asterisk/agi-bin/cid_rewrite.php on line 60
> On 5/8/05, Jay Milk <[EMAIL PROTECTED]> wrote:
Make sure the script is executable -- since it's hash-banged, you should
be able to run it from the command-line without errors. If not, chances
are you have the wrong PHP version -- this script requires 5.x, a
limitation brought on by my use of the php-agi API. When I have some
time, I'll modify this to run with all PHP versions, or provide at least
a backward compatible version.
So, run it from commmand line
("/var/lib/asterisk/agi-bin/cid_rewrite.php") and see if it throws any
errors. If it just "hangs", CTRL+C will cancel it -- it's waiting for
AGI input and should work ok. If you do see errors, they should point
you to what you're missing. I'm guessing it would be a requirement such
as PHP version or pear DB.
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Re: [Asterisk-Users] CallerID name lookup AGI script
Can you give any help on debugging AGI. The CLI just shows the following:
May 8 13:23:54 VERBOSE[18102]: -- Executing
EAGI("IAX2/[EMAIL PROTECTED]/1", "cid_rewrite.php") in new stack
May 8 13:23:54 VERBOSE[18102]: -- Launched AGI Script
/var/lib/asterisk/agi-bin/cid_rewrite.php
May 8 13:23:54 VERBOSE[18102]: -- AGI Script cid_rewrite.php
completed, returning 0
No rows are put in the db. Somewhere my setup is breaking but not sure where.
Thanks
On 5/8/05, Jay Milk <[EMAIL PROTECTED]> wrote:
> Mine does business lookups properly, and also uses a mySQL database to
> cache results (and allows you to store your own results):
>
> http://www.muware.com/asterisk/
>
> And yes, it does both google and 411.com, and falls back to telcodata.us
> to get CO information if the other two come back empty. Since it's
> already on there, it stores the address in the DB as well, if google or
> 411 return it.
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Re: [Asterisk-Users] *HOWTO* : using mime-construct with outlook - send fax to email recipient
Can anybody maybe help me a little here. I made the changes below... but I don't think I did them exactly correct. When I get a fax now, it is named: _var_spool_asterisk_fax_1114530379.5.tif.pdf I think the problem is with the second line. $p .= cont "Content-Type: $type;\n name=" . token_quote($arg) . "\n" Thanks On 2/7/05, Asterisk <[EMAIL PROTECTED]> wrote: > We've managed to setup spandsp to receive faxes and email them to the > appropriate person. > > We did all of our testing using Thunderbird, and the attached pdf files > worked very well. However, when we went "live", some people complained > that the attachments in outlook were named <>.dat instead (for > example) of 123456.pdf > > Having looked into the mime-construct script, we noticed a subtle > difference between working Outlook attachments and Thunderbird. The > differences are shown below in case anyone is interested. > > left is the original mime-construct, right is our modified version > > 332c332 > < $part_header .= "Content-Disposition: attachment; filename=" > --- > > $part_header .= "Content-Disposition: inline; filename=" > 405c405 > < $p .= cont "Content-Type: $type\n" > --- > > $p .= cont "Content-Type: $type;\n name=" . > token_quote($arg) . "\n" > > Outlook now names the attachment correctly, and Thunderbird also seems > to work just as well with the changes. > > Hope someone finds this useful. > > Julian. > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CallerID name lookup AGI script
Jim,
Thanks for sharing this. I am currently using cidlookup.agi written by
James Golovich. http://asterisk.gnuinter.net/
However the problem I have with that script and probably this one also
is that my provider sends the number as +16105551212 so I need a way
to strip out the leading two characters. Right now his just removes
everything non-numeric but that still leaves me with 16105551212 which
fails on lookups:
#remove everything non numeric from callerid string
$callerid =~ s/[^\d]//g;
I know nothing about scripting in perl. Can you offer any assistance
in how to strip the 2 leading characters? His callerid script is part
of the asterisk-perl package at http://asterisk.gnuinter.net/ ---
might want to take a look at how he does his URL queries for info as
well if you haven't already.
I really like the additional npa/nxx lookup that you are doing. Maybe
combining your efforts with his script would be helpful. He also
cache's the lookup to a directory to improve performance.
On Apr 9, 2005 3:04 PM, Jim Meehan <[EMAIL PROTECTED]> wrote:
> Hi all,
>
> My VoIP provider (race.com) doesn't send name info with CallerID, so I wrote
> an AGI script that does the following:
>
> 1) If it's a toll free number (800|888|877|866), set the CallerID name to
> "TollFree Caller"
> 2) Use curl to look up the number in Google phonebook
> 3) If a business listing, set the CallerID name to business name, as is.
> 4) If it's a residential listing, reverse the listing so it's last name first,
> then set the CallerID name to that.
> 5) If there's no match in Google phonebook, look up the NPA/NXX on
> www.areacodedownload.com and set the CallerID name to "@ST RATECENTER" where
> "ST" is the two-letter state abbreviation, and "RATECENTER" is the name of
> telco rate center in that state.
>
> Thought some of you might find this AGI script useful, so I'm including it
> below. It requires the Asterisk::AGI perl module.
>
> There are other reverse phone lookup sources that are more complete than
> Google's, but they are harder to screen scrape. Also, I probably could have
> made this a little cleaner if I used the Google API rather than screen
> scraping with curl/perl. Please feel free to take a shot at making any of
> those modifications.
>
> Here's a snippet from my extensions.conf where it gets called:
>
> exten => s,1,AGI(callerid.agi|${CALLERIDNUM})
> exten => s,2,SetCallerId,"${googlename} <${CALLERIDNUM}>"
> exten => s,3,Dial(${PHONES},30,r)
> exten => s,4,Answer
> exten => s,5,Wait(2)
> exten => s,6,Voicemail(u3001)
> exten => s,7,Hangup
>
> And here's the script:
>
> #!/usr/bin/perl
>
> use Asterisk::AGI;
>
> $AGI = new Asterisk::AGI;
>
> $number = $ARGV[0];
>
> if ($number =~ m/(800|888|877|866)\d{7}/) {
> $AGI->set_variable('googlename', "\"TollFree Caller\"");
> exit 0;
> }
>
> open(RESULTS, "/usr/bin/curl -s -m 2 -A Mozilla/4.0
> http://www.google.com/search
> ?q=phonebook:$number |");
>
> while () {
> if (m/Residential Phonebook/) {
> $reverse = 1;
> @fields = split(/>/);
> }
> if (m/Business Phonebook/) {
> @fields = split(/>/);
> }
> if (m/did not match any/) {
> @digits = split(//, $number);
> $npa = $digits[0] . $digits[1] . $digits[2];
> $nxx = $digits[3] . $digits[4] . $digits[5];
> open(LOCATION, "/usr/bin/curl -s -m 2 -A Mozilla/4.0
> http://www.areacodedown
> load.com/$npa/$nxx/ |");
> while () {
> if (m/>State $line = ;
> $line =~ m/\"\#CACACA\">\w* (\w\w)<\/td>/;
> $name = "[EMAIL PROTECTED]";
> }
> if (m/>Rate Center $line = ;
> $line =~ m/\"\#CACACA\">((\w|\s)*)<\/td>/;
> $name = $name . " " . $1;
> }
> }
> $AGI->set_variable('googlename', "\"$name\"");
> exit 0;
> }
> }
>
> @result = split(/-/, $fields[35]);
> chop($result[0]);
> if ($reverse) {
> @words = split(/ /, $result[0]);
> $last = pop(@words);
> unshift(@words, "$last,");
> foreach $word (@words) {
> $name = $name . $word . " ";
> }
> }
> if ($reverse == 0) {
> $name = $result[0];
> }
>
> $AGI->set_variable('googlename', "\"$name\"");
>
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Re: [Asterisk-Users] Any opinions on quality/service of Teliax?
Yes and yes. On Apr 9, 2005 6:03 PM, Bellows, Jared <[EMAIL PROTECTED]> wrote: > I'm looking into the TelIAX pay-as-you-go plan. I'm assuming that they > charge incoming calls minutes as well? Is there the $0.02 connection fee for > the incoming call as well? > > Thanks, > Jared ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No ringback over IAX - LiveVoip
Found this info on their website: http://www.livevoip.com/index.php?subject=2&content=networkStatus LiveVoip Operations Staff DTMF - Ringback Issues Currently, Asterisk is using the timing of the input stream to reproduce the output stream. This means that when no RTP streams are being sent from the peer Endpoint Gateway, Asterisk is unable to generate audio. This approach or limitation leads to "one way speech" conditions. Plus - Some devices don't generate audio until the answer supervision is received from the called. For all these scenarios, no ringback can be presented to the calling party. In cases where the endpoints are using silence compression, the audio from asterisk is chopped. Its fine if your run Asterisk with a T-1 Card, if not then you are going to experience issues. What Can or Should be Done? To get this solved, Asterisk should obtain its clocking from an internal source in a way that an output stream can be generated without getting any RTP input. The clocking should then be taken from an internal timing mechanism that keeps track of the synchronization. The solution should not require T1 connectivity [IE: no TDM hardware]. Such T1 connectivity would severely limit traffic on the LiveVoip Global SIP network via IP. Developers should work to solve the no alerting scenario's [when peer is set in RCV only mode] and all issues related to the use of silence compression. A configuration option should exist to choose the timing method for customers that want to use Asterisk in calling card applications or any application where no T-1 cards will ever be required. Status: LiveVoip engineers have developed a workaround for our internal switch network. This will be tested and could take up to 14 days to install in every LiveVoip Network Node location. On Tue, 15 Mar 2005 17:07:53 -0500, Robert Webb <[EMAIL PROTECTED]> wrote: > > On Tue, 15 Mar 2005 14:50:38 -0700 > Daniel Webb <[EMAIL PROTECTED]> wrote: > > On Fri, Mar 11, 2005 at 11:50:01PM +, Jay Milk > >wrote: > > > >> Dude, where have you been? This has been discussed here > >>at length. > >> Everyone agrees that it's on LiveVOIP's end, but they're > >>shrugging their > >> shoulders and pointing toward *. Search the list. > > > > Could you point out the best way to "search the list"? > > > > Perhaps go to > >http://lists.digium.com/pipermail/asterisk-users/, go to > > each month one at a time, then click "threads", then do > >a page search? > > What a swell interface. > > How about learning a few Google skills and in the search > line type: > > site:lists.digium.com > > THe above site command will only search the url specified. > In this case the Asterisk lists. > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FWD to Vonage not working?
The FWD -> Vonage interconnect has been down for some time now. Vonage claimed there was a secuity issue and pulled the plug. No word when/if it will ever be working again. > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of cmisip > Sent: Domingo, 20 de Marzo de 2005 03:39 p.m. > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] FWD to Vonage not working? > > I am having trouble with this. > > I can dial 1800 numbers fine > as well as FWD service numbers but not Vonage. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TAPI
A free solution would be to use YAC in conjunction with netcat. A guide is on the wiki. On Sun, 20 Mar 2005 15:42:07 -0600, Anton Krall <[EMAIL PROTECTED]> wrote: > OK, the outbound problem is fixed... Now, my other question is, anybody > using identapop for popup CID on your screen? > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall > Sent: Domingo, 20 de Marzo de 2005 03:34 p.m. > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: [Asterisk-Users] TAPI > > I just installed tapi and some app called identapop pro. I havent tested > incoming calls yet but so far, I cant get calls out using outlooks. > > I configured TAPI for asterisk inside outlooks and I set TAPI to these > configs: > > TAPI connects using the manager to asterisk without problems. > > As channels I configure this: > User channel: SIP/myphone and the phone actually rings when I tell outlook > to dial out. > Inbound channel: Zap/g1 but I havent tested this yet, please let me know if > it looks ok? I have a group defined as 1 wichi has 2 x100p cards. > Dial by context with nothing defined > Outbound channel: I use to have Zap/g1 but nothing happens, then I tried > forcing it to Zap/2 but this is what asterisk debug shows on the console: > > voip*CLI> > -- Lauching Dial(ZAP/20445513208717) on SIP/intruder1-2a89 Mar 20 > 15:23:58 NOTICE[18155]: app_dial.c:936 dial_exec_full: Unable to create > channel of type 'ZAP' (cause 0) > == Everyone is busy/congested at this time (1:0/0/1) > > So.. Anything looks bad? > > Thx Guys! > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Has anybody tried NVFaxDetect Fax detection SIP/IAX
No. But I would be interested in seeing how well it handles fax detection over SIP/IAX. On Mon, 14 Mar 2005 11:56:20 -0700, Joseph <[EMAIL PROTECTED]> wrote: > Has anybody tried NVFaxDetect Fax detection for sip SIP/IAX channel? > > There is a new application from Newman Telecom for fax detection. > http://www.sineapps.com/news.php?rssid=575 > > Current Asterisk Fax detection doesn't work for me as I don't have > Digium cards; I'm using Siupra > > -- > #Joseph > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Voicepulse Open Access & Asterisk Problems
I got this working if anyone out there is looking to do the same. See:
http://www.dslreports.com/forum/remark,12899866~mode=flat#12899866
After some more experimenting, I discovered that you MUST use the long
register statement ala Broadvoice. Unlike Broadvoice the service has
been ROCK SOLID. Too bad you must have a regular account first :(
On Thu, 17 Feb 2005 19:03:39 -0500, Brian Dingman <[EMAIL PROTECTED]> wrote:
> I can't seem to dial out with Voicepulse Open Access service using *.
> Incoming works fine. Another user posted a few weeks back that they
> were having problems and there are some threads at dslreports.com
> about this as well. Maybe someone here can figure out what the issue
> is from the sip debug info below. I am at a loss.
>
> The audible error message from Allison is 0984 (from VP server)
>
> Here is all the pertinent info:
>
> [sip.conf]
>
> [general]
> port = 5060
> bindaddr = 0.0.0.0
> srvlookup=yes
> tos=lowdelay
> maxexpirey=3600
> disallow=all
> allow=ulaw
> musicclass=default
> language=en
> relaxdtmf=yes
> ;useragent=Asterisk PBX
> ;nat=yes
>
> register => s00**:[EMAIL PROTECTED]
>
> externip=asterisk.briandingman.com
> localnet=192.168.1.0/255.255.0.0
>
> [voicepulse]
> type=friend
> context=voicepulse-incoming
> username=s00**
> secret=
> host=access1.voicepulse.com
> dtmf=inband
> nat=yes
> qualify=yes
> canreinvite=no
> insecure=very
>
> [1000]
> type=friend
> host=dynamic
> ;callerid=Brian <1000>
> dtmfmode=rfc2833
> mailbox=1000
> context=Home
> ;nat=no
> ;qualify=yes
> secret=
>
> Error message from CLI:
> -- Executing Macro("SIP/1000-fbdb", "vp-dial|16109951010") in new stack
> -- Executing Dial("SIP/1000-fbdb", "SIP/[EMAIL PROTECTED]") in new stack
> -- Called [EMAIL PROTECTED]
> -- SIP/voicepulse-e009 is making progress passing it to SIP/1000-fbdb
> Feb 17 17:08:42 WARNING[8523]: chan_sip.c:6811 handle_response:
> Forbidden - wrong password on authentication for INVITE to '"1000"
> ;tag=as3e632d2a'
> -- SIP/voicepulse-e009 is circuit-busy
> == Everyone is busy/congested at this time
> -- Executing Hangup("SIP/1000-fbdb", "") in new stack
> == Spawn extension (macro-vp-dial, s, 2) exited non-zero on
> 'SIP/1000-fbdb' in macro 'vp-dial'
> == Spawn extension (Home, 16109951010, 1) exited non-zero on 'SIP/1000-fbdb'
> -- Got SIP response 481 "Call Leg Does Not Exist" back from 66.234.228.159
>
> (Sorry for the length)
> SIP Debug info:
>
> -- Executing Macro("SIP/1000-cd47", "vp-dial|16109951010") in new stack
> -- Executing Dial("SIP/1000-cd47", "SIP/[EMAIL PROTECTED]") in new stack
> We're at 68.163.52.50 port 15640
> Answering/Requesting with root capability 0x4 (ulaw)
> Answering with non-codec capability 0x1 (telephone-event)
> 12 headers, 10 lines
> Reliably Transmitting:
> INVITE sip:[EMAIL PROTECTED] SIP/2.0
> Via: SIP/2.0/UDP 68.163.52.50:5060;branch=z9hG4bK600a4321;rport
> From: "1000" ;tag=as74c56bff
> To:
> Contact:
> Call-ID: [EMAIL PROTECTED]
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Date: Thu, 17 Feb 2005 22:10:02 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Content-Type: application/sdp
> Content-Length: 214
>
> v=0
> o=root 8523 8523 IN IP4 68.163.52.50
> s=session
> c=IN IP4 68.163.52.50
> t=0 0
> m=audio 15640 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> (NAT) to 66.234.228.159:5060
> -- Called [EMAIL PROTECTED]
> asterisk*CLI>
>
> Sip read:
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/UDP
> 68.163.52.50:5060;branch=z9hG4bK600a4321;received=68.163.52.50;rport=50210
> From: "1000" ;tag=as74c56bff
> To: ;tag=as1ecc3219
> Call-ID: [EMAIL PROTECTED]
> CSeq: 102 INVITE
> User-Agent: VoicePulse SW
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact:
> Proxy-Authenticate: Digest realm="uasw001.voicepulse.com", nonce="5d626333"
> Content-Length: 0
>
> 11 headers, 0 lines
> Transmitting:
> ACK sip:[EMAIL PROTECTED] SIP/2.0
> Via: SIP/2.0/UDP 68.163.52.50:5060;branch=z9hG4bK600a4321;rport
> From: "1000" ;tag=as74c56bff
> To: ;tag=as1ecc3219
> Contact:
> Call-ID: [EMAIL PROTECTED]
> CSeq: 102 ACK
> User-Agent: Asterisk PBX
> Content-Length: 0
>
> (NAT) to 66.234.228.159:5060
> We're at 68.163.52.50 port 15640
> Answering/Requesting with root capability 0x4 (ulaw
Re: [Asterisk-Users] Broadvoice Busy Issue
You get this when you lose registration. Try qualify=100 or qualify=yes, to see if that alleviates the problem. I can make and receive calls for about 30 seconds before this happens. On Mon, 14 Mar 2005 15:26:19 -0500, Randy Johnson <[EMAIL PROTECTED]> wrote: > I have my broadvoice asterisk server up and running. For some reason > after every couple of days you call the number and it says the number > you are trying to reach is busy and cannot take your call right now. > > I then stop asterisk and start it and it is fine for a couple days. > > Has anyone else had this issue? Any idea why? > > Randy > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk and Broadvoice Outgoing Again :(
I thought this patch was added into the 1.04 and later source code?
On Mon, 14 Mar 2005 07:24:44 +0200, Dimitris Kounalakis
<[EMAIL PROTECTED]> wrote:
> I never managed to make outgoing calls to broadvoice without the
> following patch to the file channels/chan_sip.c
> it comes from http://edvina.net/broadvoice/ and it is the only fraction
> that it is still needed for outgoing calls.
> It does not cause any problems with other sip devices that are connected
> to my asterisk box.
> if you do not patch it, then in sip debug you will notice that
> broadvoice gives you an error message:
> I do not remember it anymore, but it should be unauthorised or access
> not allowed something like this.
>
> --- channels/chan_sip.c.old 2005-03-12 18:10:49.0 +0200
> +++ channels/chan_sip.c 2005-03-14 07:20:18.0 +0200
> @@ -3701,16 +3701,28 @@
> /* If we have full contact, trust it */
> strncpy(invite, p->fullcontact, sizeof(invite) - 1);
> /* Otherwise, use the username while waiting for registration */
> - } else if (!ast_strlen_zero(p->username)) {
> - if (ntohs(p->sa.sin_port) != DEFAULT_SIP_PORT) {
> - snprintf(invite, sizeof(invite),
> "sip:[EMAIL PROTECTED]:%d",p->username, p->tohost, ntohs(p->sa.sin_port));
> +} else {
> + /* If we have set the fromdomain, this is also used
> + as the to domain for SIP calls to a peer. Fromdomain
> + is used for calls to SIP proxys mostly
> + */
> + char fromdomain[256];
> + if (!ast_strlen_zero(p->fromdomain)) {
> + strncpy(fromdomain, p->fromdomain,
> sizeof(fromdomain) -1);
> } else {
> - snprintf(invite, sizeof(invite),
> "sip:[EMAIL PROTECTED]",p->username, p->tohost);
> + strncpy(fromdomain, p->tohost,
> sizeof(fromdomain) -1);
> + }
> + if (!ast_strlen_zero(p->username)) {
> + if (ntohs(p->sa.sin_port) != DEFAULT_SIP_PORT) {
> + snprintf(invite, sizeof(invite),
> "sip:[EMAIL PROTECTED]:%d",p->username, fromdomain, ntohs(p->sa.sin_port));
> + } else {
> + snprintf(invite, sizeof(invite),
> "sip:[EMAIL PROTECTED]",p->username, fromdomain);
> + }
> + } else if (ntohs(p->sa.sin_port) != DEFAULT_SIP_PORT) {
> + snprintf(invite, sizeof(invite), "sip:%s:%d",
> fromdomain, ntohs(p->sa.sin_port));
> + } else {
> + snprintf(invite, sizeof(invite), "sip:%s",
> fromdomain);
> }
> - } else if (ntohs(p->sa.sin_port) != DEFAULT_SIP_PORT) {
> - snprintf(invite, sizeof(invite), "sip:%s:%d", p->tohost,
> ntohs(p->sa.sin_port));
> - } else {
> - snprintf(invite, sizeof(invite), "sip:%s", p->tohost);
> }
> strncpy(p->uri, invite, sizeof(p->uri) - 1);
> /* If there is a VXML URL append it to the SIP URL */
>
> ___
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Re: [Asterisk-Users] asterisk and Broadvoice Outgoing Again :(
Did you ever get arounnd this issue? I am seeing the same thing,
On Sun, 13 Mar 2005 00:04:54 +0545, Vicky Shrestha
<[EMAIL PROTECTED]> wrote:
> Thanks,
>
> I have that already in my /etc/hosts
>
> But it's still not working :(
>
> On Saturday 12 March 2005 03:48, Rich Adamson wrote:
> > For everyone that's trying to get BV to work, you'all might want to
> > edit your /etc/hosts file and insert something like:
> >
> > 147.135.8.128 sip.broadvoice.com
> >
> > This was a requirement from way back and I've since discontinuted
> > BV for a different provider, but seems as though of all the suggestions
> > posted in recent weeks, few mention the above.
> >
> > After editing /etc/hosts, there is no need to reboot, etc. The contents
> > are read dynamically. Then make sure that your contexts and extensions.conf
> > use sip.broadvoice.com in them. They did have four different servers
> > at one time (with four different IP's), but if you stick with one
> > (like the above) and play with the other parameters to get it to work,
> > then you can change servers at a later time.
> >
> > As one more comment, any changes that you make to sip.conf or
> > extensions.conf associated with trying to make BV work, don't forget
> > to "stop" and restart asterisk. Don't rely on a reload as it does not
> > reread all parameter changes.
> >
> >
> >
> > > I can't make outgoing calls via Broadvoice. I have tried each and every
> > > configuration that was posted to list previously.
> > >
> > > I am able to receive incoming calls fine.
> > >
> > > I get the following in asterisk console:
> > > =
> > > asterisk*CLI> show version
> > > Asterisk CVS-HEAD-03/10/05-22:51:28 built by [EMAIL PROTECTED] on a i686
> > > running Linux
> > > asterisk*CLI>
> > > -- Executing Dial("SIP/502-c147", "SIP/[EMAIL PROTECTED]") in new
> > > stack -- Called [EMAIL PROTECTED]
> > > -- Got SIP response 400 "Bad request" back from 147.135.8.128
> > > -- SIP/-19dd is circuit-busy
> > > == Everyone is busy/congested at this time (1:0/1/0)
> > > -- Executing Congestion("SIP/502-c147", "5") in new stack
> > > == Spawn extension (vicky, 0018086749157, 2) exited non-zero on
> > > 'SIP/502-c147'
> > > -- Got SIP response 400 "Bad request" back from 147.135.8.128
> > > -- Executing Dial("SIP/502-8efd", "SIP/[EMAIL PROTECTED]") in new
> > > stack -- Called [EMAIL PROTECTED]
> > > -- Got SIP response 400 "Bad request" back from 147.135.8.128
> > > -- SIP/-4bf5 is circuit-busy
> > > == Everyone is busy/congested at this time (1:0/1/0)
> > > -- Executing Congestion("SIP/502-8efd", "5") in new stack
> > > == Spawn extension (vicky, 008086749157, 2) exited non-zero on
> > > 'SIP/502-8efd'
> > > -- Got SIP response 400 "Bad request" back from 147.135.8.128
> > > -- Got SIP response 481 "Unknown Dialog" back from 147.135.8.128
> > >
> > >
> > > Here is my sip.conf:
> > > ===
> > > register =>
> > > [EMAIL PROTECTED]::[EMAIL PROTECTED]/broadvoi
> > >ce
> > >
> > > []
> > > type=peer
> > > user=phone
> > > host=sip.broadvoice.com
> > > fromdomain=sip.broadvoice.com
> > > fromuser=
> > > secret=
> > > username=
> > > insecure=very
> > > context=default
> > > authname=
> > > dtmfmode=inband
> > > dtmf=inband
> > > canreinvite=no
> > >
> > >
> > >
> > > --
> > > With regards,
> > >
> > > Vicky Shrestha
> > > System Director
> > > WorldLink Communications
> > > Jawalakhel , Kathmandu, Nepal
> > > ___
> > > Asterisk-Users mailing list
> > > Asterisk-Users@lists.digium.com
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > To UNSUBSCRIBE or update options visit:
> > >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > ---End of Original Message-
>
> --
> With regards,
>
> Vicky Shrestha
> System Director
> WorldLink Communications
> Jawalakhel , Kathmandu, Nepal
> ___
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Re: DISREGARD!![Asterisk-Users] Broadvoice outgoing problems
I doubt that was the problem. I would be interested in hearing what else you did besides that to get it working. On Sat, 12 Mar 2005 17:46:58 -0500, Jay Carter <[EMAIL PROTECTED]> wrote: > ... I just tried again after removing my hosts file entry (again) and > outbound is now working! I had taken it out before, but I think I was > getting a different error at the time. > > Sometimes it seems like asking for help is itself a cure! > > Thanks anyway! > JDC > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sending traffic to LiveVoip
You have to wait till you get an email from them saying your account
is setup. I had the same problem where my DID was setup before my
outgoing account.
On Sat, 19 Feb 2005 13:16:15 -0800, Ed Greenberg <[EMAIL PROTECTED]> wrote:
> I have several DIDs (working well) with LiveVoip and I just signed up for
> some outbound minutes. Unfortunately they did not send connection
> instructions.
>
> I tried:
> exten =>
> _1NXXNXX,2,Dial(IAX2/userid:[EMAIL PROTECTED]/${EXTEN}|60|s)
>
> but I get the error
> Feb 19 15:14:09 WARNING[21453]: chan_iax2.c:5546 socket_read: Call rejected
> by 217.160.244.186: No authority found
> -- Hungup 'IAX2/217.160.244.186:4569/1'
>
> If I change the dialing string to
> exten =>
> _1NXXNXX,2,Dial(SIP/userid:[EMAIL PROTECTED]/${EXTEN}|60|s)
>
> I get:
> Feb 19 15:15:18 WARNING[21453]: chan_sip.c:1398 create_addr: No such host:
> 217.160.244.186/14082098516
> Feb 19 15:15:18 NOTICE[21453]: app_dial.c:749 dial_exec: Unable to create
> channel of type 'SIP'
>
> Unfortunately, LiveVoip does not reliably answer technical support
> questions on the weekend. I have one in, but no response as yet.
>
> Is anybody sending traffic to LiveVoip, and what is your dialing string?
>
> (remember to edit out your user id and password :)
>
>
> ___
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[Asterisk-Users] Voicepulse Open Access & Asterisk Problems
I can't seem to dial out with Voicepulse Open Access service using *.
Incoming works fine. Another user posted a few weeks back that they
were having problems and there are some threads at dslreports.com
about this as well. Maybe someone here can figure out what the issue
is from the sip debug info below. I am at a loss.
The audible error message from Allison is 0984 (from VP server)
Here is all the pertinent info:
[sip.conf]
[general]
port = 5060
bindaddr = 0.0.0.0
srvlookup=yes
tos=lowdelay
maxexpirey=3600
disallow=all
allow=ulaw
musicclass=default
language=en
relaxdtmf=yes
;useragent=Asterisk PBX
;nat=yes
register => s00**:[EMAIL PROTECTED]
externip=asterisk.briandingman.com
localnet=192.168.1.0/255.255.0.0
[voicepulse]
type=friend
context=voicepulse-incoming
username=s00**
secret=
host=access1.voicepulse.com
dtmf=inband
nat=yes
qualify=yes
canreinvite=no
insecure=very
[1000]
type=friend
host=dynamic
;callerid=Brian <1000>
dtmfmode=rfc2833
mailbox=1000
context=Home
;nat=no
;qualify=yes
secret=
Error message from CLI:
-- Executing Macro("SIP/1000-fbdb", "vp-dial|16109951010") in new stack
-- Executing Dial("SIP/1000-fbdb", "SIP/[EMAIL PROTECTED]") in new stack
-- Called [EMAIL PROTECTED]
-- SIP/voicepulse-e009 is making progress passing it to SIP/1000-fbdb
Feb 17 17:08:42 WARNING[8523]: chan_sip.c:6811 handle_response:
Forbidden - wrong password on authentication for INVITE to '"1000"
;tag=as3e632d2a'
-- SIP/voicepulse-e009 is circuit-busy
== Everyone is busy/congested at this time
-- Executing Hangup("SIP/1000-fbdb", "") in new stack
== Spawn extension (macro-vp-dial, s, 2) exited non-zero on
'SIP/1000-fbdb' in macro 'vp-dial'
== Spawn extension (Home, 16109951010, 1) exited non-zero on 'SIP/1000-fbdb'
-- Got SIP response 481 "Call Leg Does Not Exist" back from 66.234.228.159
(Sorry for the length)
SIP Debug info:
-- Executing Macro("SIP/1000-cd47", "vp-dial|16109951010") in new stack
-- Executing Dial("SIP/1000-cd47", "SIP/[EMAIL PROTECTED]") in new stack
We're at 68.163.52.50 port 15640
Answering/Requesting with root capability 0x4 (ulaw)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 10 lines
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 68.163.52.50:5060;branch=z9hG4bK600a4321;rport
From: "1000" ;tag=as74c56bff
To:
Contact:
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 17 Feb 2005 22:10:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 214
v=0
o=root 8523 8523 IN IP4 68.163.52.50
s=session
c=IN IP4 68.163.52.50
t=0 0
m=audio 15640 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
(NAT) to 66.234.228.159:5060
-- Called [EMAIL PROTECTED]
asterisk*CLI>
Sip read:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
68.163.52.50:5060;branch=z9hG4bK600a4321;received=68.163.52.50;rport=50210
From: "1000" ;tag=as74c56bff
To: ;tag=as1ecc3219
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: VoicePulse SW
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Proxy-Authenticate: Digest realm="uasw001.voicepulse.com", nonce="5d626333"
Content-Length: 0
11 headers, 0 lines
Transmitting:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 68.163.52.50:5060;branch=z9hG4bK600a4321;rport
From: "1000" ;tag=as74c56bff
To: ;tag=as1ecc3219
Contact:
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
(NAT) to 66.234.228.159:5060
We're at 68.163.52.50 port 15640
Answering/Requesting with root capability 0x4 (ulaw)
Answering with non-codec capability 0x1 (telephone-event)
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 68.163.52.50:5060;branch=z9hG4bK709030d1;rport
From: "16109951010" ;tag=as74c56bff
To:
Contact:
Call-ID: [EMAIL PROTECTED]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Proxy-Authorization: Digest username="s00**",
realm="uasw001.voicepulse.com", algorithm=MD5,
uri="sip:[EMAIL PROTECTED]", nonce="5d626333",
response="HASH***", opaque=""
Date: Thu, 17 Feb 2005 22:10:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 214
v=0
o=root 8523 8524 IN IP4 68.163.52.50
s=session
c=IN IP4 68.163.52.50
t=0 0
m=audio 15640 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
(NAT) to 66.234.228.159:5060
asterisk*CLI>
Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
68.163.52.50:5060;branch=z9hG4bK709030d1;received=68.163.52.50;rport=50210
From: "16109951010" ;tag=as74c56bff
To: ;tag=as0630cede
Call-ID: [EMAIL PROTECTED]
CSeq: 103 INVITE
User-Agent: VoicePulse SW
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0
10 headers, 0 lines
asterisk*CLI>
Sip read:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP
68.163.52.50:5060;branch=z9hG4bK709030d1;received=6
Re: [Asterisk-Users] Asterisk and Voice Pulse Open Access
Chris,
Did you ever get this working?
On Sat, 15 Jan 2005 03:18:01 -0500, Chris Wallace
<[EMAIL PROTECTED]> wrote:
> I have researched my issue a little more and this is what I have come up
> with. Here a examples of my configurations so far and the error I get when
> I try to dial an external number. It seems like I am so close, thanks for
> the help so far!
>
> Chris
>
>
>
> ftmy-voip-01*CLI>
> -- Executing Dial("SIP/100-9c8f", "SIP/[EMAIL PROTECTED]|30|r") in
> new stack
> -- Called [EMAIL PROTECTED]
> -- SIP/voicepulse-out-a68a is making progress passing it to SIP/100-9c8f
> Jan 15 02:08:13 WARNING[17333]: chan_sip.c:6811 handle_response: Forbidden -
> wrong password on authentication for INVITE to '"Chris Wallace"
> ;tag=as772f7e09'
> -- SIP/voicepulse-out-a68a is circuit-busy
> == Everyone is busy/congested at this time
> Jan 15 02:08:19 WARNING[17333]: chan_sip.c:694 retrans_pkt: Maximum retries
> exceeded on call [EMAIL PROTECTED] for seqno 103
> (Non-critical Request)
> Jan 15 02:08:23 WARNING[17333]: pbx.c:1934 ast_pbx_run: Timeout, but no rule
> 't' in context 'local'
> ftmy-voip-01*CLI>
>
>
>
>
>
> ;
> ; SIP Configuration for Asterisk
> ;
> [general]
> port=5060
> bindaddr=0.0.0.0
> context=default
> externip=69.138.121.16
>
> register => s00**:[EMAIL PROTECTED]
>
> [voicepulse-out]
> type=peer
> context=voicepulse-out
> username=s00**
> authuser=s00**
> secret=
> host=access1.voicepulse.com
> nat=yes
>
> [voicepulse-in]
> type=friend
> context=vp-incoming
> username=s00**
> secret=
> host=access1.voicepulse.com
> nat=yes
>
> [100]
> type=friend
> context=local
> username=100
> secret=1234
> callerid="Chris Wallace" <239-935-0299>
> host=dynamic
> nat=yes
> canreinvite=no
>
>
>
>
>
> ;
> ; Extension Configuration for Asterisk
> ;
> [general]
> static=yes
> writeprotect=no
>
> [globals]
>
> [vp-incoming]
> exten => 2399350299,1,Answer
> exten => 2399350299,2,Wait,1
> exten => 2399350299,3,Playback(vm-goodbye)
> exten => 2399350299,4,Hangup
>
> [local]
> exten => _9X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
> include=internal
>
> [internal]
> exten => 100,1,Dial(SIP/100,20)
> exten => 100,2,Voicemail(u100)
> exten => 100,102,Voicemail(b100)
>
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Randy
> Sent: Friday, January 14, 2005 11:30 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Asterisk and Voice Pulse Open Access
>
> Chris,
>
> I do not have VoicePulse Open Access, but I do have an incoming number
> through
> VoicePulse Connect. You might want to give the following a try unless you
> get
> a repsonse back from someone who uses Open Access specifically. (I found
> the
> access1.voicepulse.com address from another posting.)
>
> Edit sip.conf and extensions.conf as follows, editing the 2165551212 to
> match your assigned phone number from VoicePulse, as well as filling in your
> userid and password.
>
> To have the extension go to another context than default, you must specify
> it
> as the context in the general section in sip.conf - I was unable to get the
> normal peer matching to work for voicepulse, at the moment I suspect its due
> to inconsistent rev mappings for their ip's. If you do not have an
> extension
> that matches your number, it will defer to 's'.
>
> sip.conf
>
> ; in your [general] section add:
> register => userid:[EMAIL PROTECTED]
>
> extensions.conf
>
> ; add an extension matching your phone number to your default context (or
> the
> ; context specified in sip.conf)
> exten => 2165551212,1,Answer
> exten => 2165551212,2,Wait,1
> exten => 2165551212,3,Playback(vm-goodbye)
> exten => 2165551212,4,Hangup
>
> Hope this works for you - it does for me with VoicePulse Connect.
>
> Randy
>
> On Fri, Jan 14, 2005 at 10:19:17PM -0500, Chris Wallace wrote:
> >
> >Has any messed with getting Asterisk to work using the Voice Pulse
> >Open Access plan? I currently have 2 numbers with Voi
Re: [Asterisk-Users] Sixtel.net / IAX.CC - Vanity Toll-Free Numbe r
I hate to beat a dead horse, but it looks like their toll free offerings are completely gone. If you try and assign a random 800 number to your account, it says they are out of stock. I dont't think any of us we being seeing a number from them any time soon. Their IVR for customer support is a nice touch as well. On Tue, 15 Feb 2005 15:07:37 -0500, Andrejus Stavickis <[EMAIL PROTECTED]> wrote: > Hi, > > I've tried to make toll-free DID work for the last 2-3 weeks. Apparently > only the IAX.CC/Sixtel personnel can make a call to my toll-free. Anybody > else just gets a busy signal. It takes for them about 5-6 business days to > respond to my request. It seems they are looking in the once a week to the > High priority tickets. > > It also seems that they are permanently closed ! No matter when I call their > support 800 number, I get "you reached us during non-business hours. Our > business hours are from 10AM to 11PM Mountain Standard Time". I've tried to > call 8AM EST, 11AM EST, 2PM EST, 4PM EST, 6PM EST, 9PM EST, 11PM EST with > completely no luck ! Maybe I'm missing something ? Does somebody know exact > difference between EST/MST ? > > Completely dissatisfied ! > > Maybe some of you knows good toll-free DID provider that does not have (or > have very little like IAX/Sixtel) monthly charges ? > > Sincerely, > > --Andy > x6722 > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sixtel.net / IAX.CC - Vanity Toll-Free Number
Sam thing here. Waiting 10+ business days for my DID. Can't get through to them by phone and email responses take days. These guys are worthless. On Tue, 15 Feb 2005 10:40:44 -0500, Mark Eissler <[EMAIL PROTECTED]> wrote: > > On Feb 15, 2005, at 10:26 AM, BJ Weschke wrote: > > > I've had the same experience. I've been waiting 7+ business days for > > their "unlimited incoming minutes" DIDs which were supposed to be > > provisioned "within 1-4 hours". > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Festival Woes
Wow. I posted that a long time ago. Thanks. Festival doesn't seem very stable to me though. On Tue, 15 Feb 2005 15:14:47 +1100, Rod Bacon <[EMAIL PROTECTED]> wrote: > SIOD ERROR: wrong type of argument to car : wholeutt > > Try changing your festival.scm to the following: > > (Note the extra () on the 4th last line). > > (define (tts_textasterisk string mode) > "(tts_textasterisk STRING MODE) > Apply tts to STRING. This function is specifically designed for > use in server mode so a single function call may synthesize the string. > This function name may be added to the server safe functions." > (let ((wholeutt (utt.synth (eval (list 'Utterance 'Text string) > (utt.wave.resample wholeutt 8000) > (utt.wave.rescale wholeutt 5) > (utt.send.wave.client wholeutt))) > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX Voice Quality Issues
No quite sure what you mean by timimg source. But I have a T100P card in the box and zaptel drivers loaded. On Wed, 9 Feb 2005 13:49:15 -0500, Paul Rodan <[EMAIL PROTECTED]> wrote: > That jitter buffer has caused nothing but problems for me. > > But that was a few months ago, haven't tried it lately. > > What are you using as your timing source? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX Voice Quality Issues
Silence Suppression is off or set to no of the SPA. I changed jitterbuffer=no and things seem better. Will need to do some more testing. On Wed, 09 Feb 2005 13:03:06 -0500, Andres <[EMAIL PROTECTED]> wrote: > > > > > >Just as further info, I am using a SPA-2000 to connect to * with G711u > >as the preferred codec. > > > > > > > > > Maybe you have silence suppression enabled on the SPA? That does not > play nice with Asterisk. > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX Voice Quality Issues
I am running * 1.0.5 and have been having lots of problems with outgoing calls and their sound quality. I am using ULAW for the codec and sixtel for termination. Basically the problem is that portions of the call seem to be lost and replaced with silence. Sometimes I can't hear the person talking othertimes they can't hear me. This situation comes and goes throughout the call. Bandwidth isn't an issue as I have a 3MB/1MB connection and there is at most 2 concurrent connections. Also using pingplotter to monitor iax2.sixtel.net shows little or no packetloss. Just as further info, I am using a SPA-2000 to connect to * with G711u as the preferred codec. Anyone else experience the like or have any suggestions on what may be causing this or ideas on how to debug? Brian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SPA-841 MWI
Was does your sip.conf look like for this Sipura? On Tue, 8 Feb 2005 22:56:15 -0500 (EST), Paul Dugas <[EMAIL PROTECTED]> wrote: > Running * CVS with a SPA-841 (0.9.5) and can't seem to get the message > waiting light to come on automatically. There is a control in the web > interface to turn it on and off (seems rather curious to me but whatever). > Has anybody got an idea as to where I may be going awry? > > TIA, > > Paul > > -- > Paul A. DugasDugas Enterprises, LLC > [EMAIL PROTECTED]1711 Indian Ridge Drive > p:404-932-1355 f:770-516-4841 Woodstock, GA 30189-6856 USA > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Autodetecting faxes
Checkout http://www.voip-info.org/wiki-NVBackgroundDetect I haven't had a chance to try it yet, but supposedly it works on SIP, ZAP, and IAX. On Tue, 8 Feb 2005 21:26:28 +1100, Mike Sander <[EMAIL PROTECTED]> wrote: > That's all very well, but what do you do if you only have SIP extensions and > IAX trunk - no Zaptel card. > > Will Fax detection still work at all? > > Thanks > > Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Autodetecting faxes
Thank you. Worked like a charm. I never would have caught that on my own. On Mon, 07 Feb 2005 23:06:54 -0500, Roger Gulbranson <[EMAIL PROTECTED]> wrote: > On Mon, 2005-02-07 at 19:39 -0500, Brian Dingman wrote: > > Here is my zapata.conf file for grins > > > > [channels] > > ; > > ; X100P plugged into PSTN > > ; > > context=fromPSTN > > signalling=fxs_ks > > echocancel=yes > > echocancelwhenbridged=yes > > echotraining=yes > > relaxdtmf=yes > > rxgain=4.5 > > txgain=0.0 > > immediate=no > > busydetect=no > > callprogress=no > > musiconhold=default > > usecallerid=no > > callerid="Incoming" <000-000-> > > ;callerid=asreceived > > channel => 1 > > faxdetect=both > > The faxdetect=both must be BEFORE the channel => 1 > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail timeouts after 30sec's everytime.
No. My answer was assuming you didn't have one in originally. On Tue, 08 Feb 2005 13:46:26 +1100, David Uzzell <[EMAIL PROTECTED]> wrote: > Brian Dingman wrote: > > This is just a guess, but try an Answer before sending it to VM. > Hmm ok not sure what that would do but I am willing to try anything at > the moment. > > Here is the incomming from Extensions.conf > > [default] > exten => 61290071091,1,Wait,1 > exten => 61290071091,n,Answer > exten => 61290071091,n,DigitTimeout,3 > exten => 61290071091,n,ResponseTimeout,5 > exten => 61290071091,n,Dial(SIP/800,60) > exten => 61290071091,n,Waitexten > exten => 61290071091,n,Playback,voicemail/default/801/unavail > exten => 61290071091,n,Voicemail,801 > exten => 61290071091,n,Goto,t|1 > > I wouldn't put another answer in there before the VM or would I? > > Thanks > > David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] *HOWTO* : using mime-construct with outlook - send fax to email recipient
Thanks. I noticed this today also. And was curious what was causing this. Might want to add it to the wiki. On Mon, 07 Feb 2005 18:36:21 +, Asterisk <[EMAIL PROTECTED]> wrote: > We've managed to setup spandsp to receive faxes and email them to the > appropriate person. > > We did all of our testing using Thunderbird, and the attached pdf files > worked very well. However, when we went "live", some people complained > that the attachments in outlook were named <>.dat instead (for > example) of 123456.pdf > > Having looked into the mime-construct script, we noticed a subtle > difference between working Outlook attachments and Thunderbird. The > differences are shown below in case anyone is interested. > > left is the original mime-construct, right is our modified version > > 332c332 > < $part_header .= "Content-Disposition: attachment; filename=" > --- > > $part_header .= "Content-Disposition: inline; filename=" > 405c405 > < $p .= cont "Content-Type: $type\n" > --- > > $p .= cont "Content-Type: $type;\n name=" . > token_quote($arg) . "\n" > > Outlook now names the attachment correctly, and Thunderbird also seems > to work just as well with the changes. > > Hope someone finds this useful. > > Julian. > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Autodetecting faxes
I ran across this on the wiki"
http://www.voip-info.org/wiki-NVBackgroundDetect
Is anyone using this? Seems a little more robust.
On Mon, 7 Feb 2005 19:39:05 -0500, Brian Dingman <[EMAIL PROTECTED]> wrote:
> Here is my zapata.conf file for grins
>
> [channels]
> ;
> ; X100P plugged into PSTN
> ;
> context=fromPSTN
> signalling=fxs_ks
> echocancel=yes
> echocancelwhenbridged=yes
> echotraining=yes
> relaxdtmf=yes
> rxgain=4.5
> txgain=0.0
> immediate=no
> busydetect=no
> callprogress=no
> musiconhold=default
> usecallerid=no
> callerid="Incoming" <000-000->
> ;callerid=asreceived
> channel => 1
> faxdetect=both
>
>
> On Mon, 7 Feb 2005 19:16:18 -0500, Brian Dingman <[EMAIL PROTECTED]> wrote:
> > No, the fax extension position is at the end... Does it matter?
> >
> > show dialplan is as follows for the fax extension:
> >
> > [ Context 'fax' created by 'pbx_config' ]
> > 'h' =>1. system(/usr/local/bin/sendEmail -q -f ${FROM}
> > -t ${TO} -u ${SUBJECT} -a ${FAXFILE} -m ${MESSAGE}) [pbx_config]
> > 2. Hangup()
> > [pbx_config]
> > 's' =>1.
> > SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif) [pbx_config]
> > 2. SetVar(FROM=emailaddress) [pbx_config]
> > 3. SetVar(TO=emailaddress)[pbx_config]
> > 4. SetVar(MESSAGE=Fax is attached)
> > [pbx_config]
> > 5. SetVar(SUBJECT=New Fax Received)
> > [pbx_config]
> > 6. rxfax(${FAXFILE})
> > [pbx_config]
> >
> >
>
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Re: [Asterisk-Users] Voicemail timeouts after 30sec's everytime.
This is just a guess, but try an Answer before sending it to VM. On Tue, 08 Feb 2005 11:34:30 +1100, David Uzzell <[EMAIL PROTECTED]> wrote: > Ok I have a challange that I can't seem to find a way to fix it. > > My Voicemail in * timesout after 30secs without fail everytime no matter > what I do. > > I have incomming calls comming in through Freshtel IAX2, if it goes to > SIP extension when it is online it can hang on for what ever time the > call goes for. > > If however it goes to the Voicemail it will timeout at 30sec and I can't > seem to figure out why it just hangs up the call. > > From below is my voicemail.conf were I have tried everything I can > think of from making the maxmessage large and turning off > silencesuppresion but nothing changines the fact that it will time out > at 30sec's. > > Does any have any ideas as to why this would do this and how I could go > about correcting the issue? > > Thanks. > David > > [general] > format=wav49|gsm|wav > [EMAIL PROTECTED] > attach=yes > maxmessage=360 > minmessage=1 > maxgreet=60 > maxsilence=0 > maxlogins=3 > > sendvoicemail=yes > review=yes > > [zonemessages] > eastern=Australian/Sydney|'vm-received' Q 'digits/at' IMp > > [default] > > ;mailboxs follow here. > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Autodetecting faxes
Here is my zapata.conf file for grins
[channels]
;
; X100P plugged into PSTN
;
context=fromPSTN
signalling=fxs_ks
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
relaxdtmf=yes
rxgain=4.5
txgain=0.0
immediate=no
busydetect=no
callprogress=no
musiconhold=default
usecallerid=no
callerid="Incoming" <000-000->
;callerid=asreceived
channel => 1
faxdetect=both
On Mon, 7 Feb 2005 19:16:18 -0500, Brian Dingman <[EMAIL PROTECTED]> wrote:
> No, the fax extension position is at the end... Does it matter?
>
> show dialplan is as follows for the fax extension:
>
> [ Context 'fax' created by 'pbx_config' ]
> 'h' =>1. system(/usr/local/bin/sendEmail -q -f ${FROM}
> -t ${TO} -u ${SUBJECT} -a ${FAXFILE} -m ${MESSAGE}) [pbx_config]
> 2. Hangup() [pbx_config]
> 's' =>1.
> SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif) [pbx_config]
> 2. SetVar(FROM=emailaddress) [pbx_config]
> 3. SetVar(TO=emailaddress)[pbx_config]
> 4. SetVar(MESSAGE=Fax is attached)[pbx_config]
> 5. SetVar(SUBJECT=New Fax Received) [pbx_config]
> 6. rxfax(${FAXFILE}) [pbx_config]
>
>
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Re: [Asterisk-Users] Autodetecting faxes
No, the fax extension position is at the end... Does it matter?
show dialplan is as follows for the fax extension:
[ Context 'fax' created by 'pbx_config' ]
'h' =>1. system(/usr/local/bin/sendEmail -q -f ${FROM}
-t ${TO} -u ${SUBJECT} -a ${FAXFILE} -m ${MESSAGE}) [pbx_config]
2. Hangup() [pbx_config]
's' =>1.
SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif) [pbx_config]
2. SetVar(FROM=emailaddress) [pbx_config]
3. SetVar(TO=emailaddress)[pbx_config]
4. SetVar(MESSAGE=Fax is attached)[pbx_config]
5. SetVar(SUBJECT=New Fax Received) [pbx_config]
6. rxfax(${FAXFILE}) [pbx_config]
On Mon, 07 Feb 2005 16:35:32 -0700, Michael Welter <[EMAIL PROTECTED]> wrote:
> Adrian Chapman wrote:
>
> > Changing the order of things in extensions.conf around a smidge got it
> > all working nicely :-
> >
> > [inbound-from-pstn]
> > include => default
> > exten => s,1,Answer
> > exten => s,2,Wait,1
> > exten => s,3,Playback(thank-you-for-calling-please-wait-a-moment)
> > exten => fax,1,Macro(faxreceive)
> > exten => s,4,
> >
> Is the position of the fax extension, between priorities 3 and 4,
> significant? What does 'show dialplan' display for the fax extension?
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Fwd: [Asterisk-Users] Autodetecting faxes
Yes. It just never gets there. I have tested spandsp and associated libs by using: exten => s,2,Goto(fax,s,1) and it works fine. * just won't auto detect the fax call. On Tue, 08 Feb 2005 10:16:26 +1100, Howard Lowndes <[EMAIL PROTECTED]> wrote: > On Tue, 2005-02-08 at 09:49, Brian Dingman wrote: > > I am having the same problems. No matter what I try, * won't detect > > faxes. I have faxdetect=both in zaptel.conf and my extensions.conf > > looks like this: > > > > [fromPSTN] > > exten => s,1,Answer > > exten => s,2,DigitTimeout(2) > > exten => s,3,ResponseTimeout(10) > > exten => s,4,Wait(3) > > exten => s,5,Background(custom/ivr-greeting) > > > > exten => i,1,Wait(1) > > exten => i,2,Background(pbx-invalid) ; "That's not valid, try again" > > exten => i,3,Goto(s,5) > > > > exten => t,1,Background(goodbye) > > exten => t,2,Hangup > > > > exten => fax,1,Goto(fax,s,1) > > Apologies. I meant "You do have a context called "Fax"...?" ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Autodetecting faxes
I am having the same problems. No matter what I try, * won't detect faxes. I have faxdetect=both in zaptel.conf and my extensions.conf looks like this: [fromPSTN] exten => s,1,Answer exten => s,2,DigitTimeout(2) exten => s,3,ResponseTimeout(10) exten => s,4,Wait(3) exten => s,5,Background(custom/ivr-greeting) exten => i,1,Wait(1) exten => i,2,Background(pbx-invalid) ; "That's not valid, try again" exten => i,3,Goto(s,5) exten => t,1,Background(goodbye) exten => t,2,Hangup exten => fax,1,Goto(fax,s,1) On Sun, 06 Feb 2005 20:42:07 +, Adrian Chapman <[EMAIL PROTECTED]> wrote: > What we found was that the fax/voice decision was being made before the > intermittent "beep--beep--beep" fax tone was being generated, so > it wasn't being detected. > > Changing the order of things in extensions.conf around a smidge got it > all working nicely :- > > [inbound-from-pstn] > include => default > exten => s,1,Answer > exten => s,2,Wait,1 > exten => s,3,Playback(thank-you-for-calling-please-wait-a-moment) > exten => fax,1,Macro(faxreceive) > exten => s,4, > > The wait allows the start of the Playback to be heard by the caller - > without it, we were finding the first word clipped. That second plus the > duration of the "Thank you for calling" message gives enough time for > the roughly 2.5sec duration between fax beeps to repeat, no matter when > it last fell compared to the answer. > > We've not checked more into the three rings before answer, but there's > been discussion (here? elsewhere?) that it's down to the wait for caller > ID. Try turning that off. TBH, I *like* the three rings - as a caller, > it psychologically gives you time to get your head in gear before the > call's answered. > > Besides - If you're ringing from a mobile, it also gives you time to > physically put the phone to your ear... > > -- > Adrian Chapman > Director > Trivas Ltd > Business on the Move > Mobility - Messaging - Infrastructure - Security - Remote Access > 07796 690210 - 01582 626552 > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Call forwarding of IAX inbound call
Just for others edification. The problem here was that I was not performing an Answer before issuing the DIAL command to the forwarded number. Once I did that the calls natively bridged and left my system completely. On Sun, 6 Feb 2005 15:31:43 -0500, Brian Dingman <[EMAIL PROTECTED]> wrote: > I am trying to do the following: > 1. Call comes in to my * box over IAX (VP Connect DID) > 2. Check to see if call should be forwarded to my cell > 3. Forward the call to my cell phone and take * out of the media path. > > I am able to do all of the above except * is not able to natively > bridge the call. I am using sixtel and for the call forward portion, > but the calls don't connect before sixtel hangs up. > > -- Attempting native bridge of > IAX2/[EMAIL PROTECTED]:4569/1 and IAX2/sixtel/3 > -- Channel 'IAX2/sixtel/3' ready to transfer > -- Releasing IAX2/sixtel/3 and IAX2/[EMAIL PROTECTED]:4569/1 > -- Hungup 'IAX2/sixtel/3' > > Any thoughts on getting a native bridge between the two providers? > > In iax.conf I DO NOT have notransfer=yes anywhere and am using ULAW > for both legs. > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicepulse DNID is blank - Any other options?
A lot of times we all overlook the obvious or easiest way to do things :) On Sun, 6 Feb 2005 13:26:25 -0800 (PST), Tim Burt <[EMAIL PROTECTED]> wrote: > Ah.. the obvious. I don't know why I missed it. > > I am just a newbie at this PBX stuff. > > Thanks for the pointer. It worked. First off. > > Hopefully, someday soon, I will contribute more than silly questions to > this list! > > Thanks again! > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call status after Answer
Who is your DID provider? On Sun, 6 Feb 2005 15:03:19 -0500, Scott Simpson <[EMAIL PROTECTED]> wrote: > Hi, > I setup asterisk as an autoattendant. When I call using IAX I get the > autoattendent okay, but when I dial one of the extensions, there is no > ringing sound passed back to the caller. > > It happens when I use my DID number, but I also configured a context so I > can get it to happen with Firefly (iax client) as the caller. It seems that > once the Answer command is executed in the dialplan, status commands > (RINGING, etc) aren't passed back through the IAX channel. > > My only workaround has been to use music on hold instead of making a ringing > sound. > > Has anyone seen this, or a solution. It seems basic, but I have been > working all day on it. > > I've tried this with 1-0-2, 1-0.5 and the head version, but all behave the > same. The IAX trace shows that RINGING is getting sent back to the client. > > Thanks > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicepulse DNID is blank - Any other options?
Maybe I am missing your exact point, but what about handling this in your extensions.conf [voicepulse-incoming] exten => 2124007999,1,Goto(nyc,s,1) exten => 2124007998,1,Goto(nyc2,s,1) That will put calls to 2124007999 into context nyc and calls to 2124007998 into context nyc2. I guess the real questions is what is your ultimate goal? On Sun, 6 Feb 2005 12:52:21 -0800 (PST), Tim Burt <[EMAIL PROTECTED]> wrote: > I just signed up for a second voicepulse number. > > I assumed that I would be able to differentiate which number the caller > dialed. > > But DNID is empty and DIALEDPEERNUMBER is empty and CHANNEL contains the > same info (almost, with the exception of a randomly assigned suffix) for > both numbers. > > Does anyone know how I might determine which number was called? > > Note, this is not CALLERID. I need the number that the caller CALLED. > > As a last resort, I guess I could use a different provider for the second > number. > > Can anyone shed any light? > > Thanks in advance! > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call forwarding of IAX inbound call
I am trying to do the following: 1. Call comes in to my * box over IAX (VP Connect DID) 2. Check to see if call should be forwarded to my cell 3. Forward the call to my cell phone and take * out of the media path. I am able to do all of the above except * is not able to natively bridge the call. I am using sixtel and for the call forward portion, but the calls don't connect before sixtel hangs up. -- Attempting native bridge of IAX2/[EMAIL PROTECTED]:4569/1 and IAX2/sixtel/3 -- Channel 'IAX2/sixtel/3' ready to transfer -- Releasing IAX2/sixtel/3 and IAX2/[EMAIL PROTECTED]:4569/1 -- Hungup 'IAX2/sixtel/3' Any thoughts on getting a native bridge between the two providers? In iax.conf I DO NOT have notransfer=yes anywhere and am using ULAW for both legs. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TAPI integration with * using Identapop software
I haven't tried identapop, but an alternative is to use netcat along with YAC listener on the windows PC. See the wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk%20call%20notification Works well for me. On Sat, 5 Feb 2005 11:49:03 +, John Middleton <[EMAIL PROTECTED]> wrote: > Hi, > I've got Outlook to call the number on * using the TAPI interface > documented on the Wiki. Its working OK. > > I have downloaded the Indentapop application, and it appears to > connect to * Ok using the Debug modes, but It isnt detecting incoming > calls. > > Has anyone git identapop working? > > Care to share configuration details? > > Thanks > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE:Terrible inbound call quality vs. outbound
I have contacted VP regarding this issue and have included links to this thread. My ticket number is [Incident: 050120-92] for reference. Might want to fire off an email referencing it On Fri, 4 Feb 2005 18:27:18 -0500, Daryl G. Jurbala <[EMAIL PROTECTED]> wrote: > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of > > Gene Willingham > > Sent: Tuesday, February 01, 2005 6:49 PM > > To: asterisk-users@lists.digium.com > > Subject: [Asterisk-Users] RE:Terrible inbound call quality > > vs. outbound > > > > > > > > I am experiencing the same problem, except I do not use > > Voicepulse outbound. > > I have 100 Mbps connection, so it should not be a bandwidth > > issue. Last > > Thursday they had a 4 hour outage on inbound calls. The call > > quality has deteriorated since. I am in the process of > > looking for another provider. > [...] > > Not to just "me too", butme too. I've contacted their support on > numerous occasions, and have been given busywork to do (run ping plotter > for 24 hours, send us the results, etc) and never receive a response > that acknowledges a problem of any sort. > > Daryl > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip
I took them up on their offer for a refund. IMHO they shouldn't offer * service at all. Even outgoing calls aren't handled properly. Lots of making progress - no answer results. Others have suggested iax.cc. However, they haven't repsonded to my email (over 2 days now) and I can't get through to them over the phone or IM. Not very promising. All I want is a toll free DID that works on * and isn't too expensive. Any suggestions for a provider? I don't even care if it can be ported away! On Thu, 3 Feb 2005 10:12:02 -0500, Mark Eissler <[EMAIL PROTECTED]> wrote: > Based on the support and management responses that have been posted to > this list it doesn't sound to me (at least) like LiveVoip really wants > business from * users anyhow. They blame a lot of problems on * and are > quick to offer a refund. There are plenty of DID providers that are > more asterisk-friendly. > > -mark > > On Feb 2, 2005, at 1:25 PM, Brian Dingman wrote: > > > Finally got a reply from LV support. Not what I was hoping for. > > Hopefully they will file a bug with Digium since they investigated the > > issue not holding my breath. > > > > Since this is such basic * functionality that they can't seem to > > accomplish I would think twice before aquiring DID's from them. > > > > LiveVoip Support > > > > Our people have looked into this matter over the past few days. They > > tell me > > that it is a problem with Asterisk. > > We are not going to be able to help you with this. If you would like a > > refund so that you can migrate to another > > service provider we will be happy to do so. With each rev. of Asterisk > > more > > and more improvements are made. > > At some point these issues may resolve but, for the time being it is > > not a > > problem we can help you with. > > > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip
Finally got a reply from LV support. Not what I was hoping for. Hopefully they will file a bug with Digium since they investigated the issue not holding my breath. Since this is such basic * functionality that they can't seem to accomplish I would think twice before aquiring DID's from them. LiveVoip Support Our people have looked into this matter over the past few days. They tell me that it is a problem with Asterisk. We are not going to be able to help you with this. If you would like a refund so that you can migrate to another service provider we will be happy to do so. With each rev. of Asterisk more and more improvements are made. At some point these issues may resolve but, for the time being it is not a problem we can help you with. On Sun, 30 Jan 2005 18:15:10 -0500, Steven Frazier <[EMAIL PROTECTED]> wrote: > I just got a couple of numbers (activated Friday) from livevoip, I am having > similar issues. > > When you call the number, I get ring back, but as soon as IVR picks up, I > should here "extensioni" I don't hear that but then I dial an extension > number and there is no ring back. I don't have this issue from other voip > providers. > > Steve > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Terrible inbound call quality vs. outbound
I have also had issues with VP Connect ONLY on incoming calls also. It doesn't happen all the time and has cleared up in recent weeks. But when it happens, it would sound like I was listening to the caller through a blown speaker. Have you reported this problem to them? Some things to try would be to turn off the jitterbuffer and disable trunking. On Tue, 1 Feb 2005 16:12:45 -0800, Robert Goodyear <[EMAIL PROTECTED]> wrote: > Oh I've tried all manner of packet shaping and QoS tagging... it's > certainly not an issue with the ISP. > > I think Gene Willingham may have the right answer, that VoicePulse > cannot handle the load. > > Anyone else have any thoughts? Maybe I need to find another IAX service > provider to test a different DID in my area. > > /rg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outlook Integration
Placing a call from Outlook is trivial with AST TAPI. How does one accomplish a screen pop in Outlook? Here is some more info from fonality: PBXtray is not developed as part of Asterisk source. It is written in an entirely different language (C++) and is a Windows app. It integrates with our media gateway, which in turn talks to Asterisk. The GPL mandate states that modifications to the source code of a GPL app need to be made available to the community. PBXtra, and PBXtray, do not make a single change to Asterisk source code. On Tue, 1 Feb 2005 21:49:34 -, Bill Seddon <[EMAIL PROTECTED]> wrote: > > We've also written a system tray based call monitor that can call out using > information from Outlook (or other sources). Our application uses the > Manager API to receive from and send commands to Asterisk as does the > Nicolás Gudiño's FOP - no change to Asterisk required. I imagine PBXTray > also uses the Manager API. > > > > Bill Seddon > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outlook Integration
This is what Fonality support said about the app and the GPL: Unfortunately the PBXtray app only works with our systems, and we cannot sell it separately. It is not released under the GPL because there are no modifications to Asterisk or any related software for it to run. On Tue, 1 Feb 2005 16:05:15 -0500, Brian Dingman <[EMAIL PROTECTED]> wrote: > So I guess that begs the question... Does anybody know where to get > the PBXtray app that Fonality uses? > > What if they are using IdentaPoP for window Pop functionality. > Probably can't get that under the GPL. > > On Mon, 31 Jan 2005 12:53:23 -0800, Manjit Riat <[EMAIL PROTECTED]> wrote: > > The partner list shows digium as one of their partners. So under GPL they > > have to provide the source code for the app. > > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outlook Integration
So I guess that begs the question... Does anybody know where to get the PBXtray app that Fonality uses? What if they are using IdentaPoP for window Pop functionality. Probably can't get that under the GPL. On Mon, 31 Jan 2005 12:53:23 -0800, Manjit Riat <[EMAIL PROTECTED]> wrote: > The partner list shows digium as one of their partners. So under GPL they > have to provide the source code for the app. > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outlook Integration
Well the AST TAPI program http://www.omniis.com/ntsgr/cms/page.asp?688 works well, but no screen pops. To answer your question, basically outlook tells * to call your extension and then call the number in outlook. On Tue, 1 Feb 2005 13:47:44 -0700 (MST), Dan Adams <[EMAIL PROTECTED]> wrote: > Curiousity question, do you know how this would work, maybe as operating > as a softphone in windows? > > Dan > > On Tue, 1 Feb 2005, Brian Dingman wrote: > > > I have been looking around for Outlook Integration for Asterisk. Saw > > the Asterisk TAPI wiki page and also ran across this: > > http://www.fonality.com/pop.cgi?page=pop_pbxtray.tt (PBXtray) > > > > It looks like Fonality has managed to make an app that does screen > > pops and allows click to dial. Has anyone else been able to get this > > all to work successfully? Looks pretty slick. > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Outlook Integration
I have been looking around for Outlook Integration for Asterisk. Saw the Asterisk TAPI wiki page and also ran across this: http://www.fonality.com/pop.cgi?page=pop_pbxtray.tt (PBXtray) It looks like Fonality has managed to make an app that does screen pops and allows click to dial. Has anyone else been able to get this all to work successfully? Looks pretty slick. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk friendly VoIP providers
http://www.iax.cc has Vegas numbers. On Sun, 30 Jan 2005 17:05:57 -0800, Manjit Riat <[EMAIL PROTECTED]> wrote: > > > > I am thinking of dumping broadvoice so I need some other VoIP providers that > have a las vegas DID and a service better than broadvoice. > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip
There is the little problem of having to switch numbers and then communicating to everyone that the number has changed. This also only seems to be a problem on inbound calls. On Sat, 29 Jan 2005 23:34:49 -0500, Andrew Kohlsmith <[EMAIL PROTECTED]> wrote: > On January 29, 2005 11:29 pm, Brian Dingman wrote: > > This is driving me crazy. I have resorted to using the m option in the > > Dial command just so folks don't hang up. I can't believe nobody else > > is having this issue. > > Simple test: try it with another VOIP provider. Throw $5 at a nufone account, > or an iax.cc account. See what happens. Hell you're already saying it's > working with other providers, so what's your data showing you? > > Why do people insist on staying with VOIP providers who provide spotty > performance and half-assed answers to technical support issues? > > -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: No ringback over IAX - LiveVoip
This is driving me crazy. I have resorted to using the m option in the
Dial command just so folks don't hang up. I can't believe nobody else
is having this issue.
Any ideas to work around this?
On Wed, 26 Jan 2005 12:11:42 -0500, Brian Dingman <[EMAIL PROTECTED]> wrote:
> Some more info. Using this exact call flow, ringback works for PSTN
> callers over WIldcard, IAX Callers over VP Connect, but NOT IAX
> callers over LiveVoip. Could this possibly be a bug with their "new"
> patch?
>
>
> On Wed, 26 Jan 2005 11:43:59 -0500, Brian Dingman <[EMAIL PROTECTED]> wrote:
> > Here is the call flow:
> > [ivr-incoming]
> > exten => s,1,LookupCIDName
> > exten => s,2,DigitTimeout(2)
> > exten => s,3,ResponseTimeout(10)
> > exten => s,4,Wait(1)
> > exten => s,5,Background(custom/ivr-incoming)
> >
> > exten => 1,1,Background(pls-wait-connect-call)
> > exten => 1,2,Dial(${RINGPHONENUMBERS},20,r)
> > exten => 1,3,Voicemail,u${VMBOX}
> > exten => 1,4,Hangup
> >
> > Running * 1.0.5. The calling party hears the please wait while I
> > connect your call, but does not hear any ringing. I tried inserting
> > exten => 1,1,Ringing but that does not work either.
> >
> > The same call flow from the pstn DOES generate ringback:
> > [fromPSTN]
> > exten => s,1,DigitTimeout(2)
> > exten => s,2,ResponseTimeout(10)
> > exten => s,3,Wait(1)
> > exten => s,4,Background(custom/ivr-greeting)
> >
> > exten => 1,1,Background(pls-wait-connect-call)
> > exten => 1,2,Dial(${RINGPHONENUMBERS},15,r)
> > exten => 1,3,Voicemail,u${VMBOX}
> > exten => 1,4,Hangup
> >
> > Any thoughts.
> >
>
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Re: [Asterisk-Users] IAX outgoing redundancy
So when should you receive a NOANSWER back? Doesn't that imply you are using DIAL with a timeout value? Otherwise I can't see how you would ever get there. I agree with you about LiveVoip. They claim to be an Asterisk service provider but anytime you have a problem they tell you that asterisk is full of bugs and not their only supported platform. On Fri, 28 Jan 2005 02:21:36 -0500, Andrew Kohlsmith <[EMAIL PROTECTED]> wrote: > On January 27, 2005 11:20 pm, Brian Dingman wrote: > > To combat this problem you will want to change the following line to > > actually do something: > > exten => dial-NOANSWER,1,Hangup > > That's a *large* failure on LiveVoip's part, IMO. If I get a NOANSWER back I > don't *want* to do anything -- there was no answer so I don't want to try to > dial out again through another provider. > > I've tried pretty much every VOIP provider out there... nufone (for me) has > been the absolute best. I've *never* had any of this bullshit I'm seeing on > the list like I am with the Broadvoice and LiveVoip type providers. it just > effing works. > > -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX outgoing redundancy
Just as an fyi.. one of the problems I am having with LiveVoip and my
guess is that some of you are also is that the LiveVoip call starts
making progress but for whatever reason it comes back and says nobody
available.
To combat this problem you will want to change the following line to
actually do something:
exten => dial-NOANSWER,1,Hangup
On Sun, 23 Jan 2005 01:50:21 -0500, Andrew Kohlsmith
<[EMAIL PROTECTED]> wrote:
> My nufone dialing macro looks something like this:
> exten => _NXXNXX,1,Macro(nufone-dial,${EXTEN})
>
> [macro-nufone-dial]
> ... (skip some stuff that sets the account code for my own billing)
> exten => s,n,Dial(${NUFONEIAX}/1${ARG1},,g)
> exten => s,n,Goto(dial-${DIALSTATUS},1)
>
> exten => dial-CANCEL,1,Hangup
> exten => dial-ANSWER,1,Hangup
> exten => dial-NOANSWER,1,Hangup
> exten => dial-BUSY,1,Busy
> exten => dial-CONGESTION,1,Macro(pri-dial,${ARG1},${ARG2})
> exten => dial-CHANUNAVAIL,1,Macro(pri-dial,${ARG1},${ARG2})
>
> IMO I shouldn't have to try and use the backup on CONGESTION, only if the
> channel was unavailable.
>
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Re: [Asterisk-Users] Sound quality tuning with VOIP/Grandstreams... echo, cut out, codecs, asterisk
PCMU is g711 ULAW and PCMA is G711 ALAW. ALAW is more common in Europe. Not sure about VAD. On Thu, 27 Jan 2005 18:44:09 -0700, Kim Lux <[EMAIL PROTECTED]> wrote: > > Thanks for the tips. > > The Grandstream doesn't have a G711 or uLaw option for codecs. It has > PCMU, PCMA and iLBC. Are any of these related to G711 ? > > Grandstreams have echo cancellation and it appears to be working after a > few seconds of conversation. > > What is VAD ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone having problems with LiveVoIP?
They definitely have capacity issues as well. It is not uncommon to get a busy signal when placing a call. i.e. Making Progess - Nobody is available. This is a real pain too since the Dial Command won't rollover to the next step once it starts making progress. On Thu, 27 Jan 2005 12:59:17 -0500, Glenn Powers <[EMAIL PROTECTED]> wrote: > > I ordered an 800# from LiveVoIP two days ago. I can register with > Asterisk just fine, but when I call my 800#, I get a fast busy. I > emailed support a day and a half ago and have heard NOTHING from them. > > VoicePulse Connect and VoipJet both work great for me. > > Someone on -users said "you get what you pay for" regarding LiveVoip. > They couldn't have been more correct! > > cheers, > glenn > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone having problems with LiveVoIP?
>From Support: Asterisk is full of bugs and in many cases you fix one thing only to have another show up. We suggested users move to 1.0.3 Our team will look at more things in the software as a part of our ongoing support to clients. We are looking at this version as well as 1.0.3 for some other issues now but, Asterisk is not our only platform. In other words Asterisk isn't there only problem. On Thu, 27 Jan 2005 10:30:07 -0500, Mark Eissler <[EMAIL PROTECTED]> wrote: > If the problem is with asterisk userswhy is LiveVoip trying to > change something at their end? > > -mark > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone having problems with LiveVoIP?
Is anyone else having issues with ringback (see my other post to this list) since the "patch" last night. On Wed, 26 Jan 2005 09:33:50 -0600, Tim Lewis <[EMAIL PROTECTED]> wrote: > LiveVoIP did not issue any end user patches last night. They had a > problem connecting to Level 3's network. LiveVoIP claimed the problem > was with asterisk users, I have not upgrade or install any patches and > all is fine now. > > My main problem with LiveVoIP has been the LACK of customer service. > They don't answer the phones or responded to email in a timely manner. > How hard would it had been to post a message about the outage? > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: No ringback on IAX channel after selecting menu option
Some more info. Using this exact call flow, ringback works for PSTN
callers over WIldcard, IAX Callers over VP Connect, but NOT IAX
callers over LiveVoip. Could this possibly be a bug with their "new"
patch?
On Wed, 26 Jan 2005 11:43:59 -0500, Brian Dingman <[EMAIL PROTECTED]> wrote:
> Here is the call flow:
> [ivr-incoming]
> exten => s,1,LookupCIDName
> exten => s,2,DigitTimeout(2)
> exten => s,3,ResponseTimeout(10)
> exten => s,4,Wait(1)
> exten => s,5,Background(custom/ivr-incoming)
>
> exten => 1,1,Background(pls-wait-connect-call)
> exten => 1,2,Dial(${RINGPHONENUMBERS},20,r)
> exten => 1,3,Voicemail,u${VMBOX}
> exten => 1,4,Hangup
>
> Running * 1.0.5. The calling party hears the please wait while I
> connect your call, but does not hear any ringing. I tried inserting
> exten => 1,1,Ringing but that does not work either.
>
> The same call flow from the pstn DOES generate ringback:
> [fromPSTN]
> exten => s,1,DigitTimeout(2)
> exten => s,2,ResponseTimeout(10)
> exten => s,3,Wait(1)
> exten => s,4,Background(custom/ivr-greeting)
>
> exten => 1,1,Background(pls-wait-connect-call)
> exten => 1,2,Dial(${RINGPHONENUMBERS},15,r)
> exten => 1,3,Voicemail,u${VMBOX}
> exten => 1,4,Hangup
>
> Any thoughts.
>
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[Asterisk-Users] No ringback on IAX channel after selecting menu option
Here is the call flow:
[ivr-incoming]
exten => s,1,LookupCIDName
exten => s,2,DigitTimeout(2)
exten => s,3,ResponseTimeout(10)
exten => s,4,Wait(1)
exten => s,5,Background(custom/ivr-incoming)
exten => 1,1,Background(pls-wait-connect-call)
exten => 1,2,Dial(${RINGPHONENUMBERS},20,r)
exten => 1,3,Voicemail,u${VMBOX}
exten => 1,4,Hangup
Running * 1.0.5. The calling party hears the please wait while I
connect your call, but does not hear any ringing. I tried inserting
exten => 1,1,Ringing but that does not work either.
The same call flow from the pstn DOES generate ringback:
[fromPSTN]
exten => s,1,DigitTimeout(2)
exten => s,2,ResponseTimeout(10)
exten => s,3,Wait(1)
exten => s,4,Background(custom/ivr-greeting)
exten => 1,1,Background(pls-wait-connect-call)
exten => 1,2,Dial(${RINGPHONENUMBERS},15,r)
exten => 1,3,Voicemail,u${VMBOX}
exten => 1,4,Hangup
Any thoughts.
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Re: [Asterisk-Users] Anyone having problems with LiveVoIP?
LiveVoip has a problem with Asterisk users on versions less than 1.0.3 If you are not using that version you need to upgrade now. We have a problem with two of our carriers at their gateway related to the Asterisk users. Our staff has developed a patch that is being tested at this time. Once the patch has been approved on our testbed we will move it on to the production switch environment. We do not do upgrades like this during the hours or 9 a.m. - 7 p.m. EST due to high traffic loads. We expect to do switch updates after 7 p.m. this evening that should resolve the problems you are having. LiveVoip engineers are also looking at a DTMF problem in the Asterisk software ver. 1.0.3 which may or may not involve you. Both of these issues are Asterisk software related in nature and not LiveVoip LLC switching defects. Thank You in Advance for your understanding. This issue has been placed under a master ticket for tracking. ** When contacting LiveVoip LLC Support please provide us with the latest version of Asterisk you are using, any and all logs if necessary and as much detail regarding any problems you are having. Network Operations Team LiveVoip LLC On Tue, 25 Jan 2005 20:11:43 -0600, Tim Lewis <[EMAIL PROTECTED]> wrote: > Thanks Jeff! > > I think it's a little too late to find this info out. 3 to 4 days of no > service. I have send many emails and still awaiting a response. Reminds > me of my ILEC (QWEST) > > Do you have any info on what this patch does? > > -later > > On Tue, 2005-01-25 at 18:20, Jeff Glassman wrote: > > They are coming out with a patch for the DID problem tonight. Need to have > > Asterisk 1.0.3 > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PrivacyManager not Working
So are you saying that * does not see the callerid but it should. Is
this a possible bug in the callerid application. RIght now I am seeing
that callerid isn't recognized 100% of the time (or possibly not
transmitted) when I receive calls from VP Connect. If I do a
NoOp(${CALLERIDNUM}) on incoming calls, a lot of times it comes back
UNAVAILABLE when I know the person calling has callerid being
transmitted. Just trying to figure out if this a bug in * or with VP.
This could also be related to why PrivacyManager is not working
properly. * has received the callerid info from the carrier but
couldn't translate it for some reason, so it says that callerid is
present but then sets it to UNAVAILABLE. Just a thought.
On Tue, 25 Jan 2005 10:43:57 -0500, Joseph Finley
<[EMAIL PROTECTED]> wrote:
> Well, it is also happens with Zaptel. I've had times where CallerID
> does not pickup and prompts the user to enter their phone number. They
> will hang up and call right back and it works. This only happens about
> 15% of the time though.
>
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Re: [Asterisk-Users] LiveVoip DTMF Issues
Mark, I have the same exact settings except I moved to 1.0.5. DTMF recognition is fundamental to using *. Problems like this shouldn't happen. As for the LiveVoip DID's, the two of them I have are down and out. They were rendering fast busy signals - totally different problem than DTMF, so support pulled them to figure out what was going on. Just a guess, but maybe the problem lies in the soft switches that they use or with the CLEC's. I remember an issue with Voicepulse a while back with answer supervision. This problem was isolated to a specific CLEC and possibly even a switch. Eventually they fixed the issue. Not all DID's are created equal. In my case, the DID without the problem was from Level3 and the toll free one with the issue was from Qwest. Maybe that tells us something. My VP Connect DID' s are from ITC (239-580) and Paetec (610-994)... these work. On Tue, 25 Jan 2005 14:08:17 -0500, Mark Eissler <[EMAIL PROTECTED]> wrote: > What does it say about * and providers? Err, I dunno but the whole > issue is giving me a splitting headache! Is everyone else using g.711 > too? > > This is my setup: > > -Asterisk 1.0.2 > -IAX (currently set to trunk=no) to VPC > -codec is g.711 > -tos bits are 0x18 (low delay, high throughput) > -jitterbuffer=no > > Is my understanding correct that with IAX dtmf is always sent out of > band regardless of the codec selected? > > Question: Why do you suppose only one line is okay with LiveVoip (with > regard to DTMF)? It must be something outside of Asterisk that's > causing the problem. Voicepulse doesn't really get too specific when > they acknowledge a problem though. > > -mark > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PrivacyManager not Working
Keith,
VP Connect is having issues right now with callerid being
transmitted... as much as they don't want to believe it. Sometimes it
works, sometimes it doesn't. Maybe this is part of the problem. Does
PM not work 100% of the time for you?
On Mon, 24 Jan 2005 21:29:37 -0500, Keith O'Brien
<[EMAIL PROTECTED]> wrote:
>
>
>
> I have been having problems getting PrivacyManager to work correctly.
> Right now I am running the 1/21/05 CVS but I have been unable to get this to
> work on asterisk-stable either.
>
>
>
> You can see from the debug below that the inbound call is arriving via IAX2
> and both the CALLING NUMBER and CALLING NAME are both set as "Unavailable".
> However, PrivacyManager executes and determines that the CallerID is
> present:
>
>
>
> "-- CallerID Present: Skipping"
>
>
>
> Anyone have an idea as to why this isn't working? Bug?
>
> asterisk1*CLI>
>
> Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW
>
>Timestamp: 00011ms SCall: 00335 DCall: 0 [66.234.228.170:4569]
>
>VERSION : 2
>
>CALLED NUMBER : 7326556755
>
>CALLING NUMBER : Unavailable
> <**
>
>CALLING NAME: Unavailable
> <**
>
>LANGUAGE: en
>
>USERNAME: voicepulse-in-01
>
>FORMAT : 4
>
>CAPABILITY : 1086
>
>ADSICPE : 2
>
>DATE TIME : 171511810
>
>
>
> Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
> AUTHREQ
>
>Timestamp: 00015ms SCall: 1 DCall: 00335 [66.234.228.170:4569]
>
>AUTHMETHODS : 4
>
>CHALLENGE : 123344711
>
>USERNAME: voicepulse-in-01
>
>
>
> Rx-Frame Retry[No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass:
> AUTHREP
>
>Timestamp: 00049ms SCall: 00335 DCall: 1 [66.234.228.170:4569]
>
>RSA RESULT :
> Sc+mxi0AL1JdD4Gh3s8Y5LJ13MrLm4DNNMDkCV2a5nSwuPx9djbCr2YmJO7eoxCbrP+077fdeMhpfXo
>
>
>
> -- Accepting AUTHENTICATED call from 66.234.228.170, requested format =
> 4, actual format = 4
>
> -- Executing
> PrivacyManager("[EMAIL PROTECTED]:4569]/1", "") in new
> stack
>
> Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass:
> ACCEPT
>
>Timestamp: 00051ms SCall: 1 DCall: 00335 [66.234.228.170:4569]
>
>FORMAT : 4
>
>
>
> -- CallerID Present: Skipping
> <**
>
> -- Executing Dial("[EMAIL PROTECTED]:4569]/1",
> "SIP/5001") in new stack
>
> -- Called 5001
>
>
>
> Extensions.conf
>
> ===
>
>
>
> exten => 7326556755,1,PrivacyManager
>
> exten => 7326556755,2,DIAL(SIP/5001)
>
> exten => 7326556755,3,Voicemail(u5001)
>
> exten => 7326556755,4,Hangup
>
>
>
>
>
>
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Re: [Asterisk-Users] LiveVoip DTMF Issues
Mark, I don't know what to tell you. With my DID's from VP Connect, DTMF works fine over IAX. Even one of the lines I have with LiveVoip seems OK over IAX. The other well... it really doesn't work at all. So what does this say about * and DTMF recognition over IAX? Or the service providers? On Tue, 25 Jan 2005 07:45:08 -0600, Michael Graves <[EMAIL PROTECTED]> wrote: > On Tue, 25 Jan 2005 08:24:36 -0500, Mark Eissler wrote: > > >My assumption is that most folks trunking through Voicepulse Connect > >must be using SIP since I haven't seen this problem mentioned before. > >So my conclusion is that DTMF and SIP and VPC work fine together BUT > >then you don't get to benefit from the efficiency of IAX. > > > >So the million dollar question is: Does IAX have a problem with DTMF or > >is it just certain carriers that have problems with DTMF? > > > >-mark > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Updating Asterisk
There was discussion of this before... I thought: cvs checkout -r v1-0 would get you the latest stable version 1.0.X code On Tue, 25 Jan 2005 08:58:46 -0500, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > Follow these simple steps to update you tree : > > # cd /usr/src > # export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot > # cvs login - the password is anoncvs. > > # cvs checkout -r v1-0-5 asterisk > # cd asterisk > # make clean; make > > then, stop asterisk > > # make install > > then start asterisk > > HTH ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] LiveVoip DTMF Issues
Out of the two DID's I have with LiveVoip, one works OK. My toll-free DID is horrific. DTMF accuracy is less than 10%. On Mon, 24 Jan 2005 14:29:12 -0700, Brandon Patterson <[EMAIL PROTECTED]> wrote: > Our people are looking at this right now and have been for the past few > days. > > Use Asterisk Ver. 1.0.3 some people encounter it where others do not. > * Problem is under investigation. > > Brandon Patterson > LiveVoip LLC > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] LiveVoip DTMF Issues
I am on 1.0.3. Very frustrating to say the least. On Mon, 24 Jan 2005 15:06:29 -0500, Mark Eissler <[EMAIL PROTECTED]> wrote: > Same problem I'm having with VP Connect. Perhaps it's a question of the > version of Asterisk being run. I'm on 1.0.2. > > -mark > > On Jan 24, 2005, at 11:36 AM, Brian Dingman wrote: > > > I have a couple of DID's with LiveVoip and am having major DTMF issues > > on incoming calls. I am connecting to them through IAX using ULAW. > > When someone dials one of these DD's (from a landline) they are for > > the most part unable to navigate the IVR menu successfuly. I would say > > the failure rate is greater than 80%. For example if the caller > > presses 5 sometimes * will see the DTMF as 55 or 555 or not see it at > > all. > > > > Is there anything I can do on my end to fix this problem, or is the > > old axim you get what you pay for true? > > > > It should also be noted that I have some other DID's from other > > providers and DTMF recognition is pretty much dead on. > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > Mark Eissler, [EMAIL PROTECTED] > Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: IAX Inbound Sound Quality
The ironic thing here is that dtmf works fine for my VP Connect DID's. Go Figure. Also outgoing call quality is perfectly fine in and out for me. The ONLY issue I have is with incoming call's incoming audio not being very crisp throughout the call. I had there regualr service with this SAME number and did not have any issues like this. IMHO it looks like the problem is on their end. On Mon, 24 Jan 2005 09:57:51 -0500, Mark Eissler <[EMAIL PROTECTED]> wrote: > > On Jan 22, 2005, at 10:49 PM, Michael Graves wrote: > > > > I notice that all four of my IAX2 based termination providers send > > incomming calls in trunking mode. You can tells since the command IAX2 > > Show Registry reports all the connections to port 8617. This is > > something that is determined at their end. In trunk mode I beleive that > > the jitter buffer is not effective. > > > > IIRC the jitter buffer is currently broken in trunk mode and should be > turned off. > > http://www.voip-info.org/wiki-Asterisk+bandwidth+iax2 > > An alternative for testing is to set trunk=no in iax.conf. I've had to > do that for my VPC trunks because I've also found that outbound faxing > seems to be broken with trunking turned on (at least to VPC). > > > FWIW, I had similar problems with VPC so I switched to Sixtel.net. No > > such problems anymore. > > > > VPC must still be using quite a lot of custom code or routing their > calls in some weird way because I've found two problems with them so > far while using IAX2: > > 1) The fax problem mentioned above. > > 2) Inbound DTMF is quite broken. (They are working on a fix and said it > would be at least 30 days...but then in December they said it would > take 2 weeks...). What a drag. > > -mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] LiveVoip DTMF Issues
I have a couple of DID's with LiveVoip and am having major DTMF issues on incoming calls. I am connecting to them through IAX using ULAW. When someone dials one of these DD's (from a landline) they are for the most part unable to navigate the IVR menu successfuly. I would say the failure rate is greater than 80%. For example if the caller presses 5 sometimes * will see the DTMF as 55 or 555 or not see it at all. Is there anything I can do on my end to fix this problem, or is the old axim you get what you pay for true? It should also be noted that I have some other DID's from other providers and DTMF recognition is pretty much dead on. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] here's my IAX callthrough app and some questions about problems I have.
Did you ever get DTMF to work reliably with LiveVoip. I am having the
exact same problems.
On Mon, 17 Jan 2005 20:22:30 -0500, Jess Coburn <[EMAIL PROTECTED]> wrote:
> Hello all,
>
> What my app does is accepts a call in on a Dial-In Number (DID) via
> IAX, and then prompts the caller for the top secret password (123) and
> then authenticates the user and prompts them to dial in the number
> they'd like to call. Once they press pound after dialing in the number
> it will read it back to them, if they press pound it will attempt to
> connect via the second IAX provider, if they press star it will allow
> them enter in the number over again.
>
> Now here's the problems and questions:
>
> 1. DTMF detection seems flawed, sometimes it's dead on but alot of
> times it will see a single keypress as multiple keypresses. So I may
> press 561 but it will see 51 and all three keypresses are about
> the same length. Is this unique to my case or do you others see this
> too. I suspect it's due to either background noise or maybe
> packetloss? Any ideas on how to clean this up?
>
> 2. The only way I can get the app to fire off is if I put the
> extension mapping in as _NXXNXX,1,CMD I'd like to use s,1,CMD but
> I don't know what I'm missing here or doing wrong.
>
> Below are a copies of my extensions.conf file and my iax.conf file.
>
> Regards,
> Jess
>
> extensions.conf
> file-
>
> [general]
> static=yes
> writeprotect=no
>
> [globals]
> ${OUTGOING-NUM}=
>
> [arbitrary-in] ; <-- Should match the context you have
>; under [incoming] in iax.conf
> exten => _NXXNXX,1,Answer
> exten => _NXXNXX,2,Background(vm-password)
> exten => _NXXNXX,3,Authenticate(123)
> exten => _NXXNXX,4,Playback(beep)
> exten => _NXXNXX,5,SetVar(NR=)
> exten => _NXXNXX,6,Goto(testdtmf|s|1)
>
> ;
> ; This context is used by the sample [arbitrary-name]
> ; context above to read back each digit you press.
> ;
> [testdtmf]
> exten => s,1,SetVar(NR=)
> exten => s,2,Background(pls-entr-num-uwish2-call)
> exten => s,3,Background(and-prs-pound-whn-finished)
> exten => s,4,Background(beep)
> exten => s,5,WaitExten(10)
> exten => _x,1,SetVar(NR=${NR}${EXTEN})
> exten => _x,2,NoOp(${NR})
> exten => _x,3,Goto(testdtmf|s|5)
> exten => _#,1,Goto(verifynumber|s|1)
> exten => i,1,Goto(testdtmf|s|1)
> exten => t,1,Hangup
>
> [verifynumber]
> exten => s,1,Background(you-dialed)
> exten => s,2,SayDigits(${NR})
> exten => s,3,Background(if-correct-press)
> exten => s,4,Background(pound)
> exten => s,5,Background(otherwise-press)
> exten => s,6,Background(star)
> exten => _#,1,Background(pls-wait-connect-call)
> exten => _#,2,Dial(IAX2/[EMAIL PROTECTED]/${NR},30)
> exten => _#,3,Background(something-terribly-wrong);
> exten => _#,4,Background(goodbye)
> exten => _#,5,Hangup
> exten => _*,1,Goto(testdtmf|s|1)
>
> iax.conf file
> --
> ; iax.conf
>
> [general]
>
> ${INCOMING-USR}=SECRET-USERNAME
> ${INCOMING-PWD}=SECRET-PWD
> ${LIVEVOIP-SVR}=217.160.244.186
>
> bandwidth=high
> disallow=lpc10
> jitterbuffer=yes
> dropcount=2
> maxjitterbuffer=500
> maxexcessbuffer=80
> minexcessbuffer=10
> jittershrinkrate=1
>
> register => ${INCOMING-USR}:[EMAIL PROTECTED]
> tos=lowdelay
>
> [incoming]
> ; this is the incoming IAX provider
> type=user
> secret=ITS-SECRET
> deny=0.0.0.0/0.0.0.0
> permit=217.160.244.186/255.255.255.0
> context=arbitrary-in
>
> [outgoing]
> ;this is the outgoing IAX provider
> type=peer
> host= 216.118.117.46
> secret= ITS-SECRET
> auth=md5
> notransfer=yes
> context=default
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Re: [Asterisk-Users] Re: IAX Inbound Sound Quality
I am seeing the following asterisk*CLI> iax2 show registry 217.160.244.186:4569 usernamexx.xx.xxx.xx:4569 60 Registered 66.234.228.170:4569 username xx.xx.xxx.xx:4569 60 Registered 65.39.205.121:4569username xx.xx.xxx.xx:4569 60 Registered On Sat, 22 Jan 2005 21:49:03 -0600, Michael Graves <[EMAIL PROTECTED]> wrote: > I notice that all four of my IAX2 based termination providers send > incomming calls in trunking mode. You can tells since the command IAX2 > Show Registry reports all the connections to port 8617. This is > something that is determined at their end. In trunk mode I beleive that > the jitter buffer is not effective. > > FWIW, I had similar problems with VPC so I switched to Sixtel.net. No > such problems anymore. > > Michael > -- > Michael Graves [EMAIL PROTECTED] > Sr. Product Specialist www.pixelpower.com > Pixel Power Inc. [EMAIL PROTECTED] > > o713-861-4005 > o800-905-6412 > c713-201-1262 > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: IAX Inbound Sound Quality
Any thoughts? Could this be a jitterbuffer problem? On Fri, 21 Jan 2005 19:08:59 -0500, Brian Dingman <[EMAIL PROTECTED]> wrote: > I have a couple of DID's through VP Connect and have been having sound > quality issues on incoming calls. During the call, the calling parties > voice sometimes sound like it is crackling, in other words it is not > very crisp. I would liken it to listening to a radio with a blown > speaker. This sound defect comes and goes throughout the call. The > other person is always audible but it just isn't as crisp and clear as > when I make outgoing calls over IAX. The other party does not hear any > audio defects. > > Anybody have any suggestions on tweaking this? Or has anyone > experienced the like? > > Running * 1.0.3 on an AMD 1700 with 512 MB of RAM (Red Hat 9). I am > the only user currently on the system. I am connecting with their IAX > server using ULAW and my SIP phone is also using ULAW (Sipura 2000). > > Thanks, > Brian > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX outgoing redundancy
Did you ever figure a way around this? It would be a good time to test
since LiveVoip is having some issues today.
On Sat, 8 Jan 2005 14:44:23 -0500, Nabeel Jafferali
<[EMAIL PROTECTED]> wrote:
Hello.
I am having an issue where sometimes the cheapest provider for certain
international destinations is not always reliable in completing calls.
However, there is not problem once the call is made (i.e. no lag or echo
or anything). The way I have it set up right now (for example) for Dar
es Salaam, Tanzania is:
exten => _925522XX.,1,Dial(IAX2/livevoip/011${EXTEN:1})
exten => _925522XX.,102,Dial(IAX2/voipjet/011${EXTEN:1})
If LiveVOIP's servers are down, it use VoipJet to complete the call,
which is great. But if LiveVOIP keeps saying "progress", * waits and
eventually times out. (BTW This is not just a problem with LiveVOIP -
for some other countries where VoipJet is primary I've had similar
problems).
Are there any ways to get around this problem? Is there a way to timeout
if "ringing" doesn't happen in 5 secs (for example) and go to the backup
provider?
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[Asterisk-Users] IAX Inbound Sound Quality
I have a couple of DID's through VP Connect and have been having sound quality issues on incoming calls. During the call, the calling parties voice sometimes sound like it is crackling, in other words it is not very crisp. I would liken it to listening to a radio with a blown speaker. This sound defect comes and goes throughout the call. The other person is always audible but it just isn't as crisp and clear as when I make outgoing calls over IAX. The other party does not hear any audio defects. Anybody have any suggestions on tweaking this? Or has anyone experienced the like? Running * 1.0.3 on an AMD 1700 with 512 MB of RAM (Red Hat 9). I am the only user currently on the system. I am connecting with their IAX server using ULAW and my SIP phone is also using ULAW (Sipura 2000). Thanks, Brian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Accessing Voice mail
Kurt,
Here is a real basic setup of how the a extension can be used in
context with the rest of the dialplan. The a extension must call
VoiceMailMain NOT Voicemail or you will get your voicemail again and
not the voicemail system.
[fromPSTN]
exten => s,1,Answer
exten => s,2,Dial(${RINGPHONENUMBERS},15,r)
exten => s,3,Voicemail,u${VMBOX}
exten => s,4,Hangup
exten => a,1,VoicemailMain
exten => a,2,Hangup
On Thu, 20 Jan 2005 15:30:14 -0500, kurt x <[EMAIL PROTECTED]> wrote:
> Brain,
>
> I did what you suggested but instead of going to VoiceMailMain it
> starts the begining of
> my recorded message each time I press the "*" key.
>
> [vmail]
> exten => a,1,Voicemail(u${ext})
> exten => a,2,Hangup
>
> Kurt
>
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Re: [Asterisk-Users] Accessing Voice mail
If you put the following in your Dialplan, pressing * should break you out of voicemail and call VoiceMailMain exten => a,1,VoicemailMain,EXTEN exten => a,2,Hangup On Wed, 19 Jan 2005 11:33:23 -0500, kurt x <[EMAIL PROTECTED]> wrote: > I want to know if there is way to break out of the voicemail message. > for example: > > On my Noterl PBX when you dial you number from any where > you get your recorded voice mail message, but during the message I > press 81 and break out of that message. It then > prompts me for my PIN thus allowing me to access my message > without using the auto attendant. > > Is this possible with Comedian? > > The below page did help. > > http://www.voip-info.org/wiki-Asterisk+cmd+VoiceMailMain > > Kurt > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie question: Can't start up asterisk
It has to do with spandsp and receiving incoming faxes. This should probably be updated in the documentation. On Tue, 18 Jan 2005 17:09:04 -0800 (PST), beonice <[EMAIL PROTECTED]> wrote: > > --- Brian Dingman <[EMAIL PROTECTED]> wrote: > > > Put /usr/local/lib in /etc/ld.so.conf then run > > ldconfig. > > > > Hmm. I don't understand what that did, but THANKS, > Brian. It seems to have at least got Asterisk to > successfully run. Now I can have fun with configuring > it! > > Thanks a bunch! > > BeOnIce. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie question: Can't start up asterisk
Put /usr/local/lib in /etc/ld.so.conf then run ldconfig. >> > [app_rxfax.so]Jan 18 15:46:05 WARN > ING[7952]: loader.c:258 as > t_load_resource: libspandsp.so.0: cannot open shared > object file: No su > ch file or directory > Jan 18 15:46:05 WARNING[7952]: loader.c: > 440 load_modules: Loading module app_rxfax.so fa > iled! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] internal dial tone on password from outside
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20DISA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No sound with X100P (clone)
Can you show us the CLI output of what is happening? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Voice Pulse Open Access
This link might help: http://www.dslreports.com/forum/remark,11775216~mode=flat On Fri, 14 Jan 2005 23:29:34 -0500, Randy <[EMAIL PROTECTED]> wrote: > Chris, > > I do not have VoicePulse Open Access, but I do have an incoming number through > VoicePulse Connect. You might want to give the following a try unless you get > a repsonse back from someone who uses Open Access specifically. (I found the > access1.voicepulse.com address from another posting.) > > Edit sip.conf and extensions.conf as follows, editing the 2165551212 to > match your assigned phone number from VoicePulse, as well as filling in your > userid and password. > > To have the extension go to another context than default, you must specify it > as the context in the general section in sip.conf - I was unable to get the > normal peer matching to work for voicepulse, at the moment I suspect its due > to inconsistent rev mappings for their ip's. If you do not have an extension > that matches your number, it will defer to 's'. > > sip.conf > > ; in your [general] section add: > register => userid:[EMAIL PROTECTED] > > extensions.conf > > ; add an extension matching your phone number to your default context (or the > ; context specified in sip.conf) > exten => 2165551212,1,Answer > exten => 2165551212,2,Wait,1 > exten => 2165551212,3,Playback(vm-goodbye) > exten => 2165551212,4,Hangup > > Hope this works for you - it does for me with VoicePulse Connect. > > Randy > > On Fri, Jan 14, 2005 at 10:19:17PM -0500, Chris Wallace wrote: > > > >Has any messed with getting Asterisk to work using the Voice Pulse > >Open Access plan? I currently have 2 numbers with Voice Pulse, 1 is a > >number that is assigned to their hardware device (Sipura SPA-2000), > >the other is a Open Access number that uses SIP from any device (you > >must authenticate with them). I want to be able to use the Open > >Access number on my Asterisk server here at home with no FXO cards. I > >have having a hard time getting this to work; I have only been using > >Asterisk for about a week now. I have managed to get Asterisk working > >with a plain phone line going into an XP100. This list is an awesome > >tool, any help would be appreciated!!! > > > > > >Chris > > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fax e-mail spandsp
At the time I didn't realize it was a common error. I thought it was a problem with the Makefile. I promise to google before I post :) Anyway it works now... somewhat. tiff's are incomplete but I will have to troubleshoot more. On Wed, 12 Jan 2005 14:37:48 +1300, Matt Riddell <[EMAIL PROTECTED]> wrote: > Steve Underwood wrote: > > The answer to this problem is the same as for every other time the same > > question has been asked. > > What he means is that if you put your error into google and click > search, it will come up with the same question being asked and answered > multiple times. > > Congratulations on the Makefile! > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel IAX2 Socket Read Error
Actually the problem was in the source code. I reinstalled the 1.0.3 sources from the tarball and rebuilt. I had to do this somewhat manually as a checkout -r v1-0 from CVS wasn't downgrading the chan_iax2.c file (which was changed yesterday) and many others. Probably doing something wrong. Anyway the latest sources seem to be problematic. The machine is an AMD 1700+ with 512MB RAM. On Wed, 12 Jan 2005 14:53:00 +1300, Matt Riddell <[EMAIL PROTECTED]> wrote: > Brian Dingman wrote: > > I grabbed the latest sources from CVS yesteday and am having problems > > compiling. * v1.0.3 was running previously without issue. I tried > > checking out the older source but get the same make errors. > > > > The box is running RH 9. I am getting the following errors. Any > > thoughts on what is wrong? > > > > chan_iax2.c:7331: internal error: Segmentation fault > > Ahhha segfault in gcc is generally a bad thing (and not usually > associated with Asterisk). > > How much RAM does the box have? > > Can you do a memtest? > > -- > Cheers, > > Matt Riddell > ___ > > http://www.sineapps.com/news.php (Daily Asterisk News - html) > http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fax e-mail spandsp
I edited the makefile and asterisk builds properly, but when I go to start it, I get the following error: [app_rxfax.so]Jan 11 18:44:12 WARNING[13877]: loader.c:258 ast_load_resource: libspandsp.so.0: cannot open shared object file: No such file or directory Jan 11 18:44:12 WARNING[13877]: loader.c:440 load_modules: Loading module app_rxfax.so failed! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Channel IAX2 Socket Read Error
I grabbed the latest sources from CVS yesteday and am having problems compiling. * v1.0.3 was running previously without issue. I tried checking out the older source but get the same make errors. The box is running RH 9. I am getting the following errors. Any thoughts on what is wrong? gcc -shared -Xlinker -x -o chan_mgcp.so chan_mgcp.o gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPT EL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-v1-0-01/10/05-20:49:47\" -DINSTALL_PR EFIX=\"\" -DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk\" -DASTV ARLIBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run\" -DASTSPOOLDIR=\"/var/ spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\" -DASTCONFPATH=\"/etc/asterisk /asterisk.conf\" -DASTMODDIR=\"/usr/lib/asterisk/modules\" -DASTAGIDIR=\"/var/li b/asterisk/agi-bin\" -DBUSYDETECT_MARTIN -Wno-missing-prototypes -Wno-m issing-declarations -DZAPATA_PRI -DIAX_TRUNKING -DCRYPTO -fPIC-c -o ch an_iax2.o chan_iax2.c chan_iax2.c: In function `socket_read': chan_iax2.c:7331: internal error: Segmentation fault Please submit a full bug report, with preprocessed source if appropriate. See http://bugzilla.redhat.com/bugzilla/> for instructions. make[1]: *** [chan_iax2.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/channels' make: *** [subdirs] Error 1 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fax e-mail spandsp
Anyone care to pass on a makefile that works. This is what my makefile.rej looks like: *** *** 71,76 rm -f $(DESTDIR)$(MODULES_DIR)/app_datetime.so rm -f $(DESTDIR)$(MODULES_DIR)/app_qcall.so app_curl.so: app_curl.o $(CC) $(SOLINK) -o $@ $< $(CURLLIBS) --- 73,84 rm -f $(DESTDIR)$(MODULES_DIR)/app_datetime.so rm -f $(DESTDIR)$(MODULES_DIR)/app_qcall.so + app_rxfax.so : app_rxfax.o + $(CC) $(SOLINK) -o $@ $< -lspandsp -ltiff + + app_txfax.so : app_txfax.o + $(CC) $(SOLINK) -o $@ $< -lspandsp -ltiff + app_curl.so: app_curl.o $(CC) $(SOLINK) -o $@ $< $(CURLLIBS) On Mon, 10 Jan 2005 23:51:03 +0800, Craig Guy <[EMAIL PROTECTED]> wrote: > Yes, > > And wrote it up in the wiki - > http://www.voip-info.org/tiki-index.php?page=Asterisk%20fax look under the > heafding 'Emailing a fax based on DID'. I used LDAP but it could just as > easily be made to work with odbcget or whatever else you wanted to use. > > Craig > > - Original Message - > From: "Altus Snyman" <[EMAIL PROTECTED]> > To: "Howard Lowndes" <[EMAIL PROTECTED]> > Cc: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Monday, January 10, 2005 8:46 PM > Subject: Re: [Asterisk-Users] fax e-mail spandsp > > > Did anyone get asterisk to actually work with a fax coming in on a pri > > number and e-mail it to a user? > > > > On Mon, 2005-01-10 at 08:29, Howard Lowndes wrote: > > > On Mon, 2005-01-10 at 16:00, Altus Snyman wrote: > > > > Its still fails! > > > > > > > > [EMAIL PROTECTED] apps]# patch < apps_makefile.patch.new > > > > patching file Makefile > > > > Hunk #1 succeeded at 42 with fuzz 2 (offset -7 lines). > > > > Hunk #2 FAILED at 73. > > > > 1 out of 2 hunks FAILED -- saving rejects to file Makefile.rej > > > > > > Yep, I've just had this one, and fixed it. > > > > > > cd asterisk/apps > > > > > > Go look at Makefile.rej and lines 19 & 20 (minus the leading "+" sign) > > > are the ones that didn't make it into Makefile. If you put them in > > > manually in the correct place then it all works. > > > > > > > > > > > On Fri, 2005-01-07 at 22:08, Jim Radford wrote: > > > > > Basically the changes in the apps/Makefile have progressed while the > patch > > > > > makefile have not. Here is a current patch that works as of > CVS-HEAD-01/06/05-14:47:06 > > > > > > > > > > Regards, > > > > > Jim > > > > > > > > > > > > > > > On Fri, 7 Jan 2005, Altus Snyman wrote: > > > > > > I'm trying to install spandsp > > > > > > But when I try to patch the Makefile it gives this error > > > > > > [EMAIL PROTECTED] apps]# patch < apps_makefile.patch > > > > > > patching file Makefile > > > > > > Reversed (or previously applied) patch detected! Assume -R? [n] y > > > > > > Hunk #1 succeeded at 41 (offset -6 lines). > > > > > > Hunk #2 FAILED at 67. > > > > > > > > > > > > is it ok to go on > > > > > > > > > > > > ___ > > > > > > Asterisk-Users mailing list > > > > > > Asterisk-Users@lists.digium.com > > > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > To UNSUBSCRIBE or update options visit: > > > > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > > > > > ___ > > > > Asterisk-Users mailing list > > > > Asterisk-Users@lists.digium.com > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > To UNSUBSCRIBE or update options visit: > > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Festival Woes
I changed all the text to lower case and removed the quotes. After
doing so, I got the following error:
SIOD ERROR: wrong type of argument to car : wholeutt
Strange thing is sometimes I get the error and sometimes I don't
On Mon, 10 Jan 2005 13:54:25 -0500, Brian Dingman <[EMAIL PROTECTED]> wrote:
> Asterisk v1.0 is running on RH 9. I installed festival RPM
> (festival-1.4.2-16.i386.rpm) and edited the festival.scm file to add:
> (define (tts_textasterisk string mode)
> "(tts_textasterisk STRING MODE)
> Apply tts to STRING. This function is specifically designed for
> use in server mode so a single function call may synthesize the string.
> This function name may be added to the server safe functions."
> (let wholeutt (utt.synth (eval (list 'Utterance 'Text string)))
> (utt.wave.resample wholeutt 8000)
> (utt.wave.rescale wholeutt 5)
> (utt.send.wave.client wholeutt)))
>
> I am getting the following error in the CLI:
> -- Executing Answer("SIP/phone1-e5f4", "") in new stack
> -- Executing Festival("SIP/phone1-e5f4", "Please record your message")
> in new stack
> == Parsing '/etc/asterisk/festival.conf': Found
> Jan 10 13:34:23 WARNING[4509]: app_festival.c:444 festival_exec:
> Festival returned ER
>
> Here is what is in the festival_server.log:
> "Load server start ./festival_server.scm"
> festival port=1314
> wrapper Mon Jan 10 13:33:57 EST 2005 : USING DEFAULT CONFIGURATION
> wrapper Mon Jan 10 13:33:57 EST 2005 : waiting
> serverMon Jan 10 13:33:57 2005 : Festival server started on port 1314
> client(1) Mon Jan 10 13:34:23 2005 : accepted from localhost
> client(1) Mon Jan 10 13:34:23 2005 : disconnected
>
> extensions.conf
> ; Record Message
> exten => _*500,1,Answer
> exten => _*500,2,Festival('Please record your message')
> exten => _*500,3,Record(mymessage:gsm)
> exten => _*500,4,Festival('You said')
> exten => _*500,5,Playback(mymessage)
>
> festival.conf
> [general]
> host=asterisk
> port=1314
> usecache=yes
> cachedir=/var/lib/asterisk/festivalcache/
> festivalcommand=(tts_textasterisk "%s" 'file)(quit)\n
>
> Any ideas?
>
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[Asterisk-Users] Festival Woes
Asterisk v1.0 is running on RH 9. I installed festival RPM
(festival-1.4.2-16.i386.rpm) and edited the festival.scm file to add:
(define (tts_textasterisk string mode)
"(tts_textasterisk STRING MODE)
Apply tts to STRING. This function is specifically designed for
use in server mode so a single function call may synthesize the string.
This function name may be added to the server safe functions."
(let wholeutt (utt.synth (eval (list 'Utterance 'Text string)))
(utt.wave.resample wholeutt 8000)
(utt.wave.rescale wholeutt 5)
(utt.send.wave.client wholeutt)))
I am getting the following error in the CLI:
-- Executing Answer("SIP/phone1-e5f4", "") in new stack
-- Executing Festival("SIP/phone1-e5f4", "Please record your message")
in new stack
== Parsing '/etc/asterisk/festival.conf': Found
Jan 10 13:34:23 WARNING[4509]: app_festival.c:444 festival_exec:
Festival returned ER
Here is what is in the festival_server.log:
"Load server start ./festival_server.scm"
festival port=1314
wrapper Mon Jan 10 13:33:57 EST 2005 : USING DEFAULT CONFIGURATION
wrapper Mon Jan 10 13:33:57 EST 2005 : waiting
serverMon Jan 10 13:33:57 2005 : Festival server started on port 1314
client(1) Mon Jan 10 13:34:23 2005 : accepted from localhost
client(1) Mon Jan 10 13:34:23 2005 : disconnected
extensions.conf
; Record Message
exten => _*500,1,Answer
exten => _*500,2,Festival('Please record your message')
exten => _*500,3,Record(mymessage:gsm)
exten => _*500,4,Festival('You said')
exten => _*500,5,Playback(mymessage)
festival.conf
[general]
host=asterisk
port=1314
usecache=yes
cachedir=/var/lib/asterisk/festivalcache/
festivalcommand=(tts_textasterisk "%s" 'file)(quit)\n
Any ideas?
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