Re: [asterisk-users] Problem with callerid(dnid) and queue
Il giorno 12/mag/10, alle ore 02:59, David Backeberg ha scritto: I thought setting CallerID like that was for setting callerID on OUTBOUND calls. Why on earth would you want to override what's happening on an inbound call? What happens if you hairpin it to a local channel, using Dial(), after you override the callerID? Following some posts and the user guide of Zoiper, DNID is the number that the caller has dialed to reach the inbound extension. For outgoing calls there is no DNID... However, my purpose is to find a way to pass the modified DNID to the softphone Zoiper in order to open the right URL. I think the Dial cmd uses the type/identifier to send the call (the caller-id should be only an additional header, shouldn't it?). Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No ringtone when going from queue to dial-command
Hello list, when I sent an incoming call first to a queue and after the timeout to a dial-command, while the correspondent's phone rings there is no ringtone for the caller... So it goes like this : 1. dial(SIP/account1,20) 2. queue(myqueue20) 3. dial(SIP/account2) In step 1 there is a ringtone for the caller. In step 2 there is musiconhold (class default) for the caller. In step 3 there is silence for as lang as the phone rings... Is there an explanation why there is no ringing-tone for the caller ?? Could I resolve this (like with Playtone) ?? Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No ringtone when going from queue to dial-command
On 12/05/10 09:08, Jonas Kellens wrote: Hello list, when I sent an incoming call first to a queue and after the timeout to a dial-command, while the correspondent's phone rings there is no ringtone for the caller... So it goes like this : 1. dial(SIP/account1,20) 2. queue(myqueue20) 3. dial(SIP/account2) In step 1 there is a ringtone for the caller. In step 2 there is musiconhold (class default) for the caller. In step 3 there is silence for as lang as the phone rings... Is there an explanation why there is no ringing-tone for the caller ?? Could I resolve this (like with Playtone) ?? Jonas. You need to use the r option in the Queue command, try: queue(myqueue,r,,,20) Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Could Asterisk PHP agi be a SOAP Client?
hi, all i want to use PHP agi to do as a soap client. does php agi support this function? Thanks! -- Thanks for your supporting, have a nice day. Sucan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problem of Cannot release Channel
Dear all, using asterisk-1.4.23.1, I encountered a problem of asterisk that cannot release the channel.* * We have several of asterisk server(client ,Guest). Now channels remaining problem occurs only in the server where the number of user agent is more than 660 and where many simultaneous calling occurs. Physically, it is being released, but in programming logic, it is not being released. If we execute core show channels concise then we see that the channels is remaining in server which is not using long time. Is it the bugs of asterisk or something else? if asterisk has limitation then how many concurrent call can occur in asterisk? Or how many user agent can register in one asterisk server? Or is it the server load problem? Please let me know. We have got the channels remaining problem in the following hand set. Acrobits Softphone version 3.2.2 (iPhone) SipSimple v4.0/iPhoneOS snom300/7.1.30 Grandstream HT487 1.0.8.16 Linphone/Linphone-3.1.2 (eXosip2/unknown)...for fax Sipdroid(Linksys/PAP2-3.1.22(LS) Is there any one who knows the solution? Please help me. Thanks in advance Nahar -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No ringtone when going from queue to dial-command
I think he need use r option in Dial command, while how I understand in Queue he need musiconhold. Dial(SIP/account2,,r) Vardan Ishfaq Malik wrote: On 12/05/10 09:08, Jonas Kellens wrote: Hello list, when I sent an incoming call first to a queue and after the timeout to a dial-command, while the correspondent's phone rings there is no ringtone for the caller... So it goes like this : 1. dial(SIP/account1,20) 2. queue(myqueue20) 3. dial(SIP/account2) In step 1 there is a ringtone for the caller. In step 2 there is musiconhold (class default) for the caller. In step 3 there is silence for as lang as the phone rings... Is there an explanation why there is no ringing-tone for the caller ?? Could I resolve this (like with Playtone) ?? Jonas. You need to use the r option in the Queue command, try: queue(myqueue,r,,,20) Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail() app not available?
Hi all, I have a demo machine I'm running up on Lenny - it has the packaged Asterisk version installed (1.4.21.2+stuff). I'm trying to add an extension to leave a voicemail message, just with Voicemail(1234), which I've done before (on 1.2 at least), but it's saying no application 'Voicemail' . module show like voi shows app_voicemail.so and app_hasnewvoicemail.so loaded (I have autoload=yes in modules.conf and have noload= a bunch, but I even explicitly set load=app_voicemail.so just in case. However, core show applications like voi only lists HasVoicemail and HasNewVoicemail, with no sign of Voicemail. The wiki seems to show that it should all be included, with no sign of deprecation... Any ideas where I can look? TIA, Andrew -- Linux supports the notion of a command line or a shell for the same reason that only children read books with only pictures in them. Language, be it English or something else, is the only tool flexible enough to accomplish a sufficiently broad range of tasks. -- Bill Garrett -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with callerid(dnid) and queue
You sure it's not using the URL OPEN parameter for the very queue?
l.
2010/5/11 Carlo Dimaggio jaasmail...@gmail.com
Hi all,
In order to use the open url function of zoiper (it opens an url
based on the asterisk $callerid(dnid)), I need rewriting of the dnid.
In my dialplan I have:
exten = 1000,3,Set(CALLERID(dnid)=newdnid)
exten = 1000,4,Noop(${CALLERID(dnid)})
exten = 1000,5,Queue(test-queue)
but the callerid(dnid) shows the extension called (the member of the
test-queue) and not the newdid. I have tried also with the option
o in cmd Dial but without success.
Do you know if there is a way to obtain the newdnid?
Thanks!
Carlo
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[asterisk-users] SIP trunk between two Asterisk servers
Hi, I'm trying to setup a SIP trunk between 2 Asterisk servers on the same LAN (no NAT, no firewalls). With IAX2 all's fine but I'm unable to setup SIP. I must be missing something obvious. I followed the simple example at http://www.panoramisk.com/90/sip-trunk-with-asterisk/en/. so Asterisk server 1 (192.168.250.111) sip.conf contains: [interboxsip] type=peer host=192.168.250.112 context=mycontext Asterisk server 2 (192.168.250.112) sip.conf contains: [interboxsip] type=peer host=192.168.250.111 context=mycontext I dialed from a SIP extension (4053) in server 2 (192.168.250.112) to 3666 in server 1 (192.168.250.111) via the interboxsip SIP trunk. The call fails and according to the SIP messages it seems to be an authentication problem. What am I missing? SIP messages on 192.168.250.112 (Asterisk server 2 - transmitting call): -- Executing [3...@from-internal:2] Dial(SIP/4053-6dea, SIP/interboxsip/3666|300|rt) in new stack Audio is at 192.168.250.112 port 15850 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.250.111:5060: INVITE sip:3...@192.168.250.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK3c951a1d;rport From: device sip:4...@192.168.250.112;tag=as4d17a185 To: sip:3...@192.168.250.111 Contact: sip:4...@192.168.250.112 Call-ID: 3770a8004ce882dd3c89c1d91b5aa...@192.168.250.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 12 May 2010 09:13:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 270 v=0 o=root 20611 20611 IN IP4 192.168.250.112 s=session c=IN IP4 192.168.250.112 t=0 0 m=audio 15850 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called interboxsip/3666 --- SIP read from 192.168.250.111:5060 --- SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK3c951a1d;received=192.168.250.112;rport=5060 From: device sip:4...@192.168.250.112;tag=as4d17a185 To: sip:3...@192.168.250.111;tag=as00842b82 Call-ID: 3770a8004ce882dd3c89c1d91b5aa...@192.168.250.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=2545a5dd Content-Length: 0 - --- (10 headers 0 lines) --- Transmitting (no NAT) to 192.168.250.111:5060: ACK sip:3...@192.168.250.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK3c951a1d;rport From: device sip:4...@192.168.250.112;tag=as4d17a185 To: sip:3...@192.168.250.111;tag=as00842b82 Contact: sip:4...@192.168.250.112 Call-ID: 3770a8004ce882dd3c89c1d91b5aa...@192.168.250.112 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/interboxsip-6deb is circuit-busy SIP messages on 192.168.250.111 (Asterisk server 1 - receiving end): -- SIP read from 192.168.250.112:5060: INVITE sip:3...@192.168.250.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport From: device sip:4...@192.168.250.112;tag=as18a568d6 To: sip:3...@192.168.250.111 Contact: sip:4...@192.168.250.112 Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 12 May 2010 09:20:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO upported: replaces Content-Type: application/sdp Content-Length: 270 v=0 o=root 20611 20611 IN IP4 192.168.250.112 s=session c=IN IP4 192.168.250.112 t=0 0 m=audio 14648 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- (14 headers 13 lines) --- Using INVITE request as basis request - 328617546726e5d430538e8061771...@192.168.250.112 Sending to 192.168.250.112 : 5060 (NAT) Reliably Transmitting (NAT) to 192.168.250.112:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;received=192.168.250.112;rport=5060 From: device sip:4...@192.168.250.112;tag=as18a568d6 To: sip:3...@192.168.250.111;tag=as57a19dac Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=1327c5b6 Content-Length: 0 --- Scheduling destruction of call '328617546726e5d430538e8061771...@192.168.250.112' in 15000 ms Found user '4053' -- SIP read from 192.168.250.112:5060: ACK sip:3...@192.168.250.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport From: device sip:4...@192.168.250.112;tag=as18a568d6 To: sip:3...@192.168.250.111;tag=as57a19dac Contact:
Re: [asterisk-users] No ringtone when going from queue to dial-command
In the queue I need musiconhold indeed, so the 'r'-option is not an option here... I did not know there was an 'r'-option for the Dial-command. However, even with this 'r'-option in the Dial-command, there is no ringtone for the caller... It just stays silent. Any other ideas ? Jonas. On 05/12/2010 10:47 AM, Vardan wrote: I think he need use r option in Dial command, while how I understand in Queue he need musiconhold. Dial(SIP/account2,,r) Vardan Ishfaq Malik wrote: On 12/05/10 09:08, Jonas Kellens wrote: Hello list, when I sent an incoming call first to a queue and after the timeout to a dial-command, while the correspondent's phone rings there is no ringtone for the caller... So it goes like this : 1. dial(SIP/account1,20) 2. queue(myqueue20) 3. dial(SIP/account2) In step 1 there is a ringtone for the caller. In step 2 there is musiconhold (class default) for the caller. In step 3 there is silence for as lang as the phone rings... Is there an explanation why there is no ringing-tone for the caller ?? Could I resolve this (like with Playtone) ?? Jonas. You need to use the r option in the Queue command, try: queue(myqueue,r,,,20) Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No ringtone when going from queue to dial-command
Try so: 1. dial(SIP/account1,20) 2. queue(myqueue,,,20) 3. Ringing 4. dial(SIP/account2,,r) 20 in queue is timeout? http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial Vardan Jonas Kellens wrote: In the queue I need musiconhold indeed, so the 'r'-option is not an option here... I did not know there was an 'r'-option for the Dial-command. However, even with this 'r'-option in the Dial-command, there is no ringtone for the caller... It just stays silent. Any other ideas ? Jonas. On 05/12/2010 10:47 AM, Vardan wrote: I think he need use r option in Dial command, while how I understand in Queue he need musiconhold. Dial(SIP/account2,,r) Vardan Ishfaq Malik wrote: On 12/05/10 09:08, Jonas Kellens wrote: Hello list, when I sent an incoming call first to a queue and after the timeout to a dial-command, while the correspondent's phone rings there is no ringtone for the caller... So it goes like this : 1. dial(SIP/account1,20) 2. queue(myqueue20) 3. dial(SIP/account2) In step 1 there is a ringtone for the caller. In step 2 there is musiconhold (class default) for the caller. In step 3 there is silence for as lang as the phone rings... Is there an explanation why there is no ringing-tone for the caller ?? Could I resolve this (like with Playtone) ?? Jonas. You need to use the r option in the Queue command, try: queue(myqueue,r,,,20) Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] One way audio problem, a=sendonly and a re-invite
Hello all, I have a problem where problem with one way audio, and I think it's related to a=sendonly and a re-invite. Can anyone please assist? The scenario is as follows - We send an INVITE to a peer, and it replies with a 100 Trying, and then a 183 Session Progress message containing a=sendonly. - Asterisk plays the caller music on hold, which I believe is correct if we have an a=sendonly. - Then the peer sends a 200 OK which also has a=sendonly, and then sends a re-invite which I've copied and pasted below. - We have canreinvite=no set in sip.conf, but I'm not sure if we should be rejecting this re-invite or not because it does contain a=sendrecv. If it should be rejected what error should Asterisk return, and how can we establish two way audio? - After this re-invite Asterisk replies with a 100 Trying and then a 200 OK which contains a=recvonly. - Call is established but called party cannot hear caller. Here's the re-invite message - note that Asterisk is on port 5070: U 2010/05/05 12:47:38.139701 (peer):5060 - (asterisk):5070 INVITE sip:(called number)@(asterisk):5070 SIP/2.0. Via: SIP/2.0/UDP (peer):5060;branch=z9hG4bK2sansay7330954rdb6594. To: User sip:(called number)@(asterisk):5070;tag=as3ddcc528. From: sip:(called number)@(peer):5060;tag=sansay7330954rdb6594. Call-ID: 58eb52aa414c5e465c3c1a15603093fb@(asterisk). CSeq: 2 INVITE. Contact: sip:(called number)@(peer):5060. Max-Forwards: 69. Content-Type: application/sdp. Content-Length: 297. . v=0. o=Sansay-VSXi 188 1 IN IP4 (peer). s=Session Controller. c=IN IP4 (other unknown IP, maybe of called number?). t=0 0. m=audio 6932 RTP/AVP 18 0 8 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=ptime:20. Any help would be much appreciated! -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Have a macro update a channel variable
Hi, I wonder if anyone can help me with a macro issue I have. I need to set a variable which tells me whether a call has been authenticated properly. However this authentication is taking place inside of a macro and I don't want to use a global variable if it will apply to other channels. I've tried using _ and __ with no real success. Is there a way of having a macro update a channel variable so when the call ends I can check the variable and handle according? I can NoOp the variable in the macro prior to changing it and it shows what it should. I then change the variable to AUTH for a successful authentication and a NoOp shows the correct value again. But when the call ends the variable going back to the original value. Thanks Lee -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.30 is slow sending STDIN to AGI script
Hi,
I still think we've either got a bug in Asterisk or a bug in the
Asterisk::AGI module.
In a separate part of the dialplan we have a call to a (much simpler)
script that begins with the below code.
In the last 1000 calls, I've had a couple of extension not returned by
AGI errors from the script.
Prior to upgrading from Asterisk 1.4.23 we never saw this error in over
10 million calls.
Unless we're doing something obviously wrong here, it would seem that
there's either a bug somewhere in Asterisk or that the Asterisk Perl
modules we have are somehow incompatible with the version of Asterisk
we're now running. $Asterisk::AGI::VERSION returns 1.01 in our
installation.
Any ideas?
#!/usr/bin/perl -w
use strict;
use Asterisk::AGI;
our $AGI = new Asterisk::AGI;
my $cwd = '/var/lib/asterisk/agi-bin';
our $gatewayID;
open(STDERR,/var/log/agi_$application.err) or die Failed to redirect
STDERR;
eval
{
my $settings = require $cwd/gw_settings.pl || die Cannot load settings
from $cwd/gw_settings.pl;
$gatewayID = $settings-{'gatewayID'};
my %input = ($AGI-ReadParse());
my $dni = $input{'extension'} || die 'extension not returned by AGI';
...
--
Cheers,
Kingsley.
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Re: [asterisk-users] SIP trunk between two Asterisk servers
Hi! I'm trying to setup a SIP trunk between 2 Asterisk servers on the same LAN (no NAT, no firewalls). With IAX2 all's fine but I'm unable to setup SIP. I must be missing something obvious. Either a) set a secret and use that on both sides, or b) look at allowguest= and the default context and maybe the domain= settings, or c) use insecure=invite Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conf files vs astdb
On Tue, May 11, 2010 at 04:48:30PM +0200, Harel Cohen wrote:
Hi all,
Could someone please tell me what is the relative cost in using conf files
oppose to the astdb? Basically I need to match a name to a phone number in
order to have all users registered by name and not by number (which I
understood is not a good practice). I have 2000 users and a complex dial-plan
and server resources become an issue.
I could implement this via a context in my extensions.conf:
exten = number,1,Dial(SIP/name) ; obviously I would need to hard-code
this for every extension
- or I could do it via astdb:
exten = _XXX,1,Dial(SIP/${DB(Names/${EXTEN})})
Which method would consume fewer resources (put aside other pro's con's)?
Is there any better way of implementing this?
Would 'hints' help me out here? If yes, I would appreciate a detailed
explanation how to use it.
How often do you update? With configuration files access is cheaper but
updating is more expensive.
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[asterisk-users] Additional CDR values
Hello, I need to store some additional CDR data from the dialplan, like in example here (down of the page): http://www.voip-info.org/wiki/index.php?page=Asterisk+func+cdr However, neither CSV, nor MySQL CDRs have any of these values as the result. Can you please highlight where can I find the nescesarry values after storing them? How should MySQL `cdr` table definition should look like? Many thanks Motiejus -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need fax solution for 1.4.xx
On 05/12/2010 08:46 AM, David Backeberg wrote: On Tue, May 11, 2010 at 3:30 PM, William Stillwell (Lists) william.stillwell-li...@ablebody.net wrote: Anybody know a reliable fax solution for 1.4.30 branch? I am using PikaFax on another server and works very well (about 3000 faxes a week), but it appears they no longer offer their product to open source asterisk, only for there “WARP” appliance. NOT really looking to migrate from 1.4.x to 1.6.x So buy an asterisk appliance that supports fax, and then you can pay somebody else to do the upgrade. http://www.digium.com/en/products/appliance/ Native 1.6 fax is really quite good. It's worth reading the release notes and doing the upgrade. Does that appliance actually support FAX? The web pages don't mention it. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunk between two Asterisk servers
--- On Wed, 5/12/10, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: Either a) set a secret and use that on both sides, or b) look at allowguest= and the default context and maybe the domain= settings, or c) use insecure=invite Thanks Philipp. I'm trying option c) which is the simplest. used insecure=invite but failed with the same SIP messages. Tried also insecure=yes but the same messages show up: SIP/2.0 407 Proxy Authentication Required I had already tried a) before but did not record the SIP messages (it also failed). I haven't tried c) yet... So I'll do a) again and log the messages and then try c). Do you actually have a working SIP trunk within your LAN? If so, could you please share your settings? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help finding online training
Are there any online training courses , similar to the Asterisk fast start course, available. I would really like to take something like the fast start course , but travelling at this point is out of the question. Any help or advice appreciated. Joe jj...@yahoo.com-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunk between two Asterisk servers
Hello
Server1:
sip.conf
[interboxserver2]
type=friend
host=192.168.250.112
context=callfromserver2
disallow=all
allow=ulaw
allow=alaw
allow=g729
extensions.conf
[callfromserver2]
exten = _X.,1,Noop(Call from server2)
exten = _X.,2,Dial(SIP/${EXTEN})
exten = _X.,3,Hangup
Server2:
sip.conf
[interboxserver1]
type=friend
host=192.168.250.111
context=callfromserver1
disallow=all
allow=ulaw
allow=alaw
allow=g729
extensions.conf
[callfromserver1]
exten = _X.,1,Noop(Call from server1)
exten = _X.,2,Dial(SIP/${EXTEN})
exten = _X.,3,Hangup
Try so, I think it must work.
And also, look and delete any another records in both servers in
sip.conf about this servers settings.
Vardan
Vieri wrote:
Hi,
I'm trying to setup a SIP trunk between 2 Asterisk servers on the same LAN
(no NAT, no firewalls).
With IAX2 all's fine but I'm unable to setup SIP. I must be missing something
obvious.
I followed the simple example at
http://www.panoramisk.com/90/sip-trunk-with-asterisk/en/.
so Asterisk server 1 (192.168.250.111) sip.conf contains:
[interboxsip]
type=peer
host=192.168.250.112
context=mycontext
Asterisk server 2 (192.168.250.112) sip.conf contains:
[interboxsip]
type=peer
host=192.168.250.111
context=mycontext
I dialed from a SIP extension (4053) in server 2 (192.168.250.112) to 3666 in
server 1 (192.168.250.111) via the interboxsip SIP trunk.
The call fails and according to the SIP messages it seems to be an
authentication problem.
What am I missing?
SIP messages on 192.168.250.112 (Asterisk server 2 - transmitting call):
-- Executing [3...@from-internal:2] Dial(SIP/4053-6dea,
SIP/interboxsip/3666|300|rt) in new stack
Audio is at 192.168.250.112 port 15850
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.250.111:5060:
INVITE sip:3...@192.168.250.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK3c951a1d;rport
From: devicesip:4...@192.168.250.112;tag=as4d17a185
To:sip:3...@192.168.250.111
Contact:sip:4...@192.168.250.112
Call-ID: 3770a8004ce882dd3c89c1d91b5aa...@192.168.250.112
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 12 May 2010 09:13:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 270
v=0
o=root 20611 20611 IN IP4 192.168.250.112
s=session
c=IN IP4 192.168.250.112
t=0 0
m=audio 15850 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called interboxsip/3666
--- SIP read from 192.168.250.111:5060 ---
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.168.250.112:5060;branch=z9hG4bK3c951a1d;received=192.168.250.112;rport=5060
From: devicesip:4...@192.168.250.112;tag=as4d17a185
To:sip:3...@192.168.250.111;tag=as00842b82
Call-ID: 3770a8004ce882dd3c89c1d91b5aa...@192.168.250.112
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=2545a5dd
Content-Length: 0
-
--- (10 headers 0 lines) ---
Transmitting (no NAT) to 192.168.250.111:5060:
ACK sip:3...@192.168.250.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK3c951a1d;rport
From: devicesip:4...@192.168.250.112;tag=as4d17a185
To:sip:3...@192.168.250.111;tag=as00842b82
Contact:sip:4...@192.168.250.112
Call-ID: 3770a8004ce882dd3c89c1d91b5aa...@192.168.250.112
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/interboxsip-6deb is circuit-busy
SIP messages on 192.168.250.111 (Asterisk server 1 - receiving end):
-- SIP read from 192.168.250.112:5060:
INVITE sip:3...@192.168.250.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport
From: devicesip:4...@192.168.250.112;tag=as18a568d6
To:sip:3...@192.168.250.111
Contact:sip:4...@192.168.250.112
Call-ID: 328617546726e5d430538e8061771...@192.168.250.112
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 12 May 2010 09:20:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
upported: replaces
Content-Type: application/sdp
Content-Length: 270
v=0
o=root 20611 20611 IN IP4 192.168.250.112
s=session
c=IN IP4 192.168.250.112
t=0 0
m=audio 14648 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
--- (14 headers 13 lines) ---
Using INVITE request as basis request -
328617546726e5d430538e8061771...@192.168.250.112
Sending to 192.168.250.112 : 5060 (NAT)
Reliably Transmitting (NAT) to 192.168.250.112:5060:
SIP/2.0 407 Proxy Authentication Required
Re: [asterisk-users] Need fax solution for 1.4.xx
I will give this a shot and see how well it will work. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Tuesday, May 11, 2010 3:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Need fax solution for 1.4.xx William Stillwell (Lists) wrote: Anybody know a reliable fax solution for 1.4.30 branch? That would be HylaFAX+ along with iaxmodem http://hylafax.sourceforge.net http://iaxmodem.sourceforge.net Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need fax solution for 1.4.xx
Dual PRI using Sangoma DAHDI Anywhere from 1 to 10 faxes a minute, averaging 2000+ a week.. Zero outbound, all inbound faxing., using about 50 did numbers. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Tuesday, May 11, 2010 3:56 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Need fax solution for 1.4.xx Free Fax for Asterisk works pretty well, but not knowing your trunk (DAHDI, E1, ??) and considering this volume, I would consider something like the MyFax service. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William Stillwell (Lists) Sent: Tuesday, May 11, 2010 2:30 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Need fax solution for 1.4.xx Anybody know a reliable fax solution for 1.4.30 branch? I am using PikaFax on another server and works very well (about 3000 faxes a week), but it appears they no longer offer their product to open source asterisk, only for there WARP appliance. NOT really looking to migrate from 1.4.x to 1.6.x -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunk between two Asterisk servers
Vardan wrote:
Hello
Server1:
sip.conf
[interboxserver2]
type=friend
host=192.168.250.112
context=callfromserver2
disallow=all
allow=ulaw
allow=alaw
allow=g729
extensions.conf
[callfromserver2]
exten = _X.,1,Noop(Call from server2)
exten = _X.,2,Dial(SIP/${EXTEN})
exten = _X.,3,Hangup
Server2:
sip.conf
[interboxserver1]
type=friend
host=192.168.250.111
context=callfromserver1
disallow=all
allow=ulaw
allow=alaw
allow=g729
extensions.conf
[callfromserver1]
exten = _X.,1,Noop(Call from server1)
exten = _X.,2,Dial(SIP/${EXTEN})
exten = _X.,3,Hangup
Try so, I think it must work.
And also, look and delete any another records in both servers in
sip.conf about this servers settings.
Vardan
Vieri wrote:
Hi,
I'm trying to setup a SIP trunk between 2 Asterisk servers on the same LAN
(no NAT, no firewalls).
With IAX2 all's fine but I'm unable to setup SIP. I must be missing
something obvious.
I followed the simple example at
http://www.panoramisk.com/90/sip-trunk-with-asterisk/en/.
so Asterisk server 1 (192.168.250.111) sip.conf contains:
[interboxsip]
type=peer
host=192.168.250.112
context=mycontext
Asterisk server 2 (192.168.250.112) sip.conf contains:
[interboxsip]
type=peer
host=192.168.250.111
context=mycontext
I dialed from a SIP extension (4053) in server 2 (192.168.250.112) to 3666
in server 1 (192.168.250.111) via the interboxsip SIP trunk.
The call fails and according to the SIP messages it seems to be an
authentication problem.
What am I missing?
SIP messages on 192.168.250.112 (Asterisk server 2 - transmitting call):
-- Executing [3...@from-internal:2] Dial(SIP/4053-6dea,
SIP/interboxsip/3666|300|rt) in new stack
Audio is at 192.168.250.112 port 15850
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.250.111:5060:
INVITE sip:3...@192.168.250.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK3c951a1d;rport
From: devicesip:4...@192.168.250.112;tag=as4d17a185
To:sip:3...@192.168.250.111
Contact:sip:4...@192.168.250.112
Call-ID: 3770a8004ce882dd3c89c1d91b5aa...@192.168.250.112
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 12 May 2010 09:13:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 270
v=0
o=root 20611 20611 IN IP4 192.168.250.112
s=session
c=IN IP4 192.168.250.112
t=0 0
m=audio 15850 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called interboxsip/3666
--- SIP read from 192.168.250.111:5060 ---
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.168.250.112:5060;branch=z9hG4bK3c951a1d;received=192.168.250.112;rport=5060
From: devicesip:4...@192.168.250.112;tag=as4d17a185
To:sip:3...@192.168.250.111;tag=as00842b82
Call-ID: 3770a8004ce882dd3c89c1d91b5aa...@192.168.250.112
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=2545a5dd
Content-Length: 0
-
--- (10 headers 0 lines) ---
Transmitting (no NAT) to 192.168.250.111:5060:
ACK sip:3...@192.168.250.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK3c951a1d;rport
From: devicesip:4...@192.168.250.112;tag=as4d17a185
To:sip:3...@192.168.250.111;tag=as00842b82
Contact:sip:4...@192.168.250.112
Call-ID: 3770a8004ce882dd3c89c1d91b5aa...@192.168.250.112
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/interboxsip-6deb is circuit-busy
SIP messages on 192.168.250.111 (Asterisk server 1 - receiving end):
-- SIP read from 192.168.250.112:5060:
INVITE sip:3...@192.168.250.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport
From: devicesip:4...@192.168.250.112;tag=as18a568d6
To:sip:3...@192.168.250.111
Contact:sip:4...@192.168.250.112
Call-ID: 328617546726e5d430538e8061771...@192.168.250.112
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 12 May 2010 09:20:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
upported: replaces
Content-Type: application/sdp
Content-Length: 270
v=0
o=root 20611 20611 IN IP4 192.168.250.112
s=session
c=IN IP4 192.168.250.112
t=0 0
m=audio 14648 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
--- (14 headers 13 lines) ---
Using INVITE request as basis request -
328617546726e5d430538e8061771...@192.168.250.112
Sending to 192.168.250.112 : 5060 (NAT)
Reliably Transmitting (NAT) to
Re: [asterisk-users] SIP trunk between two Asterisk servers
And also please show your settings and logs (without debug) Vardan Vieri wrote: --- On Wed, 5/12/10, Philipp von Klitzingklitz...@pool.informatik.rwth-aachen.de wrote: Either a) set a secret and use that on both sides, or b) look at allowguest= and the default context and maybe the domain= settings, or c) use insecure=invite Thanks Philipp. I'm trying option c) which is the simplest. used insecure=invite but failed with the same SIP messages. Tried also insecure=yes but the same messages show up: SIP/2.0 407 Proxy Authentication Required I had already tried a) before but did not record the SIP messages (it also failed). I haven't tried c) yet... So I'll do a) again and log the messages and then try c). Do you actually have a working SIP trunk within your LAN? If so, could you please share your settings? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk core dumping on SendFax with FFA
Hi All, I seem to have stumbled on a bit of a problem. When trying to send a fax with Fax For Asterisk on 1.6.2.x (have tried 1.6.2.5, 1.6.2.7 and the current svn version, with FFA 1.2 I get a core dump each time. Here is an extract form the console: [May 12 22:47:09] DEBUG[22584]: app_queue.c:1084 handle_statechange: Device 'SIP/vltb-sbc01' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [May 12 22:47:09] DEBUG[22725]: pbx.c:3692 pbx_extension_helper: Launching 'Set' -- Executing [...@tbsendfax:1] Set(SIP/vltb-sbc01-, timestarted=20100512224709) in new stack [May 12 22:47:09] DEBUG[22725]: pbx.c:3692 pbx_extension_helper: Launching 'Answer' -- Executing [...@tbsendfax:2] Answer(SIP/vltb-sbc01-, ) in new stack [May 12 22:47:09] DEBUG[22725]: pbx.c:3692 pbx_extension_helper: Launching 'Set' -- Executing [...@tbsendfax:3] Set(SIP/vltb-sbc01-, ANSWERED=1) in new stack [May 12 22:47:09] DEBUG[22725]: pbx.c:3692 pbx_extension_helper: Launching 'Set' -- Executing [...@tbsendfax:4] Set(SIP/vltb-sbc01-, CALLSTATUS=0) in new stack [May 12 22:47:09] DEBUG[22725]: pbx.c:3692 pbx_extension_helper: Launching 'Set' -- Executing [...@tbsendfax:5] Set(SIP/vltb-sbc01-, FAXOPT(localstationid)=vltbfax) in new stack [May 12 22:47:09] DEBUG[22725]: res_fax.c:2120 acf_faxopt_write: channel 'SIP/vltb-sbc01-' setting FAXOPT(localstationid) to 'vltbfax' [May 12 22:47:09] DEBUG[22725]: pbx.c:3692 pbx_extension_helper: Launching 'Set' -- Executing [...@tbsendfax:6] Set(SIP/vltb-sbc01-, FAXOPT(ecm)=no) in new stack [May 12 22:47:09] DEBUG[22725]: res_fax.c:2120 acf_faxopt_write: channel 'SIP/vltb-sbc01-' setting FAXOPT(ecm) to 'no' [May 12 22:47:09] DEBUG[22725]: pbx.c:3692 pbx_extension_helper: Launching 'SendFAX' -- Executing [...@tbsendfax:7] SendFAX(SIP/vltb-sbc01-, /var/spool/asterisk/fax/campaign_70.tif) in new stack -- Channel 'SIP/vltb-sbc01-' sending FAX '/var/spool/asterisk/fax/campaign_70.tif' [May 12 22:47:09] DEBUG[22725]: channel.c:2434 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second [May 12 22:47:09] DEBUG[22725]: channel.c:2548 ast_read_generator_actions: Generator got voice, switching to phase locked mode [May 12 22:47:09] DEBUG[22725]: channel.c:2434 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [May 12 22:47:09] DEBUG[22725]: rtp.c:3878 ast_rtp_write: Ooh, format changed from unknown to alaw [May 12 22:47:09] DEBUG[22725]: rtp.c:3904 ast_rtp_write: Created smoother: format: 8 ms: 20 len: 160 [May 12 22:47:13] DEBUG[22725]: rtp.c:1240 ast_rtcp_read: Got RTCP report of 88 bytes [May 12 22:47:15] DEBUG[22587]: chan_sip.c:8223 process_sdp: Processing session-level SDP v=0... UNSUPPORTED. [May 12 22:47:15] DEBUG[22587]: chan_sip.c:8223 process_sdp: Processing session-level SDP o=- 840372135 840372136 IN IP4 125.213.160.145... UNSUPPORTED. [May 12 22:47:15] DEBUG[22587]: chan_sip.c:8223 process_sdp: Processing session-level SDP s=ENSResip... UNSUPPORTED. [May 12 22:47:15] DEBUG[22587]: chan_sip.c:8223 process_sdp: Processing session-level SDP c=IN IP4 125.213.162.37... OK. [May 12 22:47:15] DEBUG[22587]: chan_sip.c:8223 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED. [May 12 22:47:15] DEBUG[22587]: chan_sip.c:8982 process_sdp_a_image: FaxVersion: 0 [May 12 22:47:15] DEBUG[22587]: chan_sip.c:8387 process_sdp: Processing media-level (image) SDP a=T38FaxVersion:0... OK. [May 12 22:47:15] DEBUG[22587]: chan_sip.c:8959 process_sdp_a_image: T38MaxBitRate: 9600 [May 12 22:47:15] DEBUG[22587]: chan_sip.c:8387 process_sdp: Processing media-level (image) SDP a=T38MaxBitRate:9600... OK. [May 12 22:47:15] DEBUG[22587]: chan_sip.c:8956 process_sdp_a_image: MaxBufferSize:200 [May 12 22:47:15] DEBUG[22587]: chan_sip.c:8387 process_sdp: Processing media-level (image) SDP a=T38FaxMaxBuffer:200... OK. [May 12 22:47:15] DEBUG[22587]: chan_sip.c:8991 process_sdp_a_image: FaxMaxDatagram: 72 [May 12 22:47:15] DEBUG[22587]: chan_sip.c:8387 process_sdp: Processing media-level (image) SDP a=T38FaxMaxDatagram:72... OK. [May 12 22:47:15] DEBUG[22587]: chan_sip.c:9028 process_sdp_a_image: RateManagement: transferredTCF [May 12 22:47:15] DEBUG[22587]: chan_sip.c:8387 process_sdp: Processing media-level (image) SDP a=T38FaxRateManagement:transferredTCF... OK. [May 12 22:47:15] DEBUG[22587]: chan_sip.c:9035 process_sdp_a_image: UDP EC: t38UDPRedundancy [May 12 22:47:15] DEBUG[22587]: chan_sip.c:8387 process_sdp: Processing media-level (image) SDP a=T38FaxUdpEC:t38UDPRedundancy... OK. [May 12 22:47:15] DEBUG[22587]: chan_sip.c:8996 process_sdp_a_image: FillBitRemoval: 0 [May 12 22:47:15] DEBUG[22587]: chan_sip.c:8387 process_sdp: Processing media-level (image) SDP a=T38FaxFillBitRemoval:0... OK. [May 12 22:47:15] DEBUG[22587]: chan_sip.c:9007
Re: [asterisk-users] SIP trunk between two Asterisk servers
I have forget to write for outcall in extension
server1:
[calltoserver2]
exten = _X.,1,Noop(Call to server2)
exten = _X.,2,Dial(SIP/interboxserver2/${EXTEN})
exten = _X.,3,Hangup
server2:
[calltoserver1]
exten = _X.,1,Noop(Call to server1)
exten = _X.,2,Dial(SIP/interboxserver1/${EXTEN})
exten = _X.,3,Hangup
:)
Vardan
Vardan wrote:
Hello
Server1:
sip.conf
[interboxserver2]
type=friend
host=192.168.250.112
context=callfromserver2
disallow=all
allow=ulaw
allow=alaw
allow=g729
extensions.conf
[callfromserver2]
exten = _X.,1,Noop(Call from server2)
exten = _X.,2,Dial(SIP/${EXTEN})
exten = _X.,3,Hangup
Server2:
sip.conf
[interboxserver1]
type=friend
host=192.168.250.111
context=callfromserver1
disallow=all
allow=ulaw
allow=alaw
allow=g729
extensions.conf
[callfromserver1]
exten = _X.,1,Noop(Call from server1)
exten = _X.,2,Dial(SIP/${EXTEN})
exten = _X.,3,Hangup
Try so, I think it must work.
And also, look and delete any another records in both servers in
sip.conf about this servers settings.
Vardan
Vieri wrote:
Hi,
I'm trying to setup a SIP trunk between 2 Asterisk servers on the same LAN
(no NAT, no firewalls).
With IAX2 all's fine but I'm unable to setup SIP. I must be missing
something obvious.
I followed the simple example at
http://www.panoramisk.com/90/sip-trunk-with-asterisk/en/.
so Asterisk server 1 (192.168.250.111) sip.conf contains:
[interboxsip]
type=peer
host=192.168.250.112
context=mycontext
Asterisk server 2 (192.168.250.112) sip.conf contains:
[interboxsip]
type=peer
host=192.168.250.111
context=mycontext
I dialed from a SIP extension (4053) in server 2 (192.168.250.112) to 3666
in server 1 (192.168.250.111) via the interboxsip SIP trunk.
The call fails and according to the SIP messages it seems to be an
authentication problem.
What am I missing?
SIP messages on 192.168.250.112 (Asterisk server 2 - transmitting call):
-- Executing [3...@from-internal:2] Dial(SIP/4053-6dea,
SIP/interboxsip/3666|300|rt) in new stack
Audio is at 192.168.250.112 port 15850
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.250.111:5060:
INVITE sip:3...@192.168.250.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK3c951a1d;rport
From: devicesip:4...@192.168.250.112;tag=as4d17a185
To:sip:3...@192.168.250.111
Contact:sip:4...@192.168.250.112
Call-ID: 3770a8004ce882dd3c89c1d91b5aa...@192.168.250.112
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 12 May 2010 09:13:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 270
v=0
o=root 20611 20611 IN IP4 192.168.250.112
s=session
c=IN IP4 192.168.250.112
t=0 0
m=audio 15850 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called interboxsip/3666
--- SIP read from 192.168.250.111:5060 ---
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.168.250.112:5060;branch=z9hG4bK3c951a1d;received=192.168.250.112;rport=5060
From: devicesip:4...@192.168.250.112;tag=as4d17a185
To:sip:3...@192.168.250.111;tag=as00842b82
Call-ID: 3770a8004ce882dd3c89c1d91b5aa...@192.168.250.112
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=2545a5dd
Content-Length: 0
-
--- (10 headers 0 lines) ---
Transmitting (no NAT) to 192.168.250.111:5060:
ACK sip:3...@192.168.250.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK3c951a1d;rport
From: devicesip:4...@192.168.250.112;tag=as4d17a185
To:sip:3...@192.168.250.111;tag=as00842b82
Contact:sip:4...@192.168.250.112
Call-ID: 3770a8004ce882dd3c89c1d91b5aa...@192.168.250.112
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/interboxsip-6deb is circuit-busy
SIP messages on 192.168.250.111 (Asterisk server 1 - receiving end):
-- SIP read from 192.168.250.112:5060:
INVITE sip:3...@192.168.250.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport
From: devicesip:4...@192.168.250.112;tag=as18a568d6
To:sip:3...@192.168.250.111
Contact:sip:4...@192.168.250.112
Call-ID: 328617546726e5d430538e8061771...@192.168.250.112
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 12 May 2010 09:20:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
upported: replaces
Content-Type: application/sdp
Content-Length: 270
v=0
o=root 20611 20611 IN IP4 192.168.250.112
s=session
c=IN IP4 192.168.250.112
t=0 0
m=audio 14648 RTP/AVP 0 8 101
Re: [asterisk-users] Help finding online training
I would doubt that anything you could do online (other than working with one of the on-line asterisk providers), could match the experience of going to the site and working with equipment. Jared Smith could provide a much better answer, since this is what he does. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph Schwartz Sent: Wednesday, May 12, 2010 7:57 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Help finding online training Are there any online training courses , similar to the Asterisk fast start course, available. I would really like to take something like the fast start course , but travelling at this point is out of the question. Any help or advice appreciated. Joe jj...@yahoo.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunk between two Asterisk servers
Hi! I'm trying option c) which is the simplest. used insecure=invite but failed with the same SIP messages. Tried also insecure=yes but the same messages show up: SIP/2.0 407 Proxy Authentication Required Then you have another entry in sip.conf that uses the same IP address. Delete that, or change the port on one of them, and adjust insecure= accordingly. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need fax solution for 1.4.xx
On Wed, May 12, 2010 at 7:45 AM, Steve Underwood ste...@coppice.org wrote: On 05/12/2010 08:46 AM, David Backeberg wrote: So buy an asterisk appliance that supports fax, and then you can pay somebody else to do the upgrade. Does that appliance actually support FAX? The web pages don't mention it. It occurred to me that it might not, after I wrote that. It does support fax pass-through to the onboard RJ-11 FXS ports, but it doesn't mention native softfax termination. My best answer is 'I don't know', but the absence of mention seems significant. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk core dumping on SendFax with FFA
On 05/12/2010 08:12 AM, Ben Dinnerville wrote: [May 12 22:47:15] ERROR[22725]: res_fax_digium.c:2114 dgm_fax_start: FAX handle 0: failed to queue document '/var/spool/asterisk/fax/campaign_70.tif' [May 12 22:47:15] ERROR[22725]: res_fax.c:834 generic_fax_exec: channel 'SIP/vltb-sbc01-' FAX session '0' failure, reason: 'failed to start FAX session' This is a known bug, that is fixed in the FFA 1.2.1 release which is in testing right now; if you contact Digium Support, they should be able to get you a copy of it. The issue seems to be with the error - failed to queue document '/var/spool/asterisk/fax/campaign_70.tif' However I can confirm that a) the file exists, b) it is worl readable and c) is created from a pdf with ghostscipt in the recommended fashion - gs -q -dNOPAUSE -dBATCH -sDEVICE=tiffg4 -sPAPERSIZE=a4 -sColorMode=mono -sOutputFile=campaign_70.tif combo2010.pdf It would be helpful if you could provide that file to the support technician so it can be investigated. Anyone else having core dump issues or fax failure issues with 1.2.0? This one has kept me up for 2 days now - if I had any hair i would be pulling it out now. Like I said, it's a known problem, and the fix should be out within a day or two. It was reported to us about a week ago, so if you had contacted the support department, it's likely they would have been able to shortcut your hair-pulling experience :-) -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need fax solution for 1.4.xx
On 05/12/2010 08:22 AM, David Backeberg wrote: On Wed, May 12, 2010 at 7:45 AM, Steve Underwood ste...@coppice.org wrote: On 05/12/2010 08:46 AM, David Backeberg wrote: So buy an asterisk appliance that supports fax, and then you can pay somebody else to do the upgrade. Does that appliance actually support FAX? The web pages don't mention it. It occurred to me that it might not, after I wrote that. It does support fax pass-through to the onboard RJ-11 FXS ports, but it doesn't mention native softfax termination. My best answer is 'I don't know', but the absence of mention seems significant. No, the Asterisk Appliance 50 has no onboard FAX support; it will allow audio FAX passthrough and T.38 FAX passthrough, but that's all. It has very limited onboard storage in any case, and there's not really a practical way to easily move TIFF/PDF files on and off of it, so it wouldn't make a very good device to provide FAX termination and origination without some work on the web interface. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunk between two Asterisk servers
Hi again! --- SIP read from 192.168.250.111:5060 --- SIP/2.0 407 Proxy Authentication Required You need to run the SIP debug on 192.168.250.111 to learn more about WHY the 407 is issued. Have a close look and you are likely to understand it right away. Also: Do not forget the reload after applying changes to sip.conf. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need fax solution for 1.4.xx
El 12/05/10 08:22, David Backeberg escribió: On Wed, May 12, 2010 at 7:45 AM, Steve Underwoodste...@coppice.org wrote: On 05/12/2010 08:46 AM, David Backeberg wrote: So buy an asterisk appliance that supports fax, and then you can pay somebody else to do the upgrade. Does that appliance actually support FAX? The web pages don't mention it. It occurred to me that it might not, after I wrote that. It does support fax pass-through to the onboard RJ-11 FXS ports, but it doesn't mention native softfax termination. My best answer is 'I don't know', but the absence of mention seems significant. Take a look at http://sourceforge.net/projects/agx-ast-addons/ There is fax support for 1.4 inside that modules. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57 1)6500816 Móvil: (+57)3138873587 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunk between two Asterisk servers
--- On Wed, 5/12/10, Vardan hvarda...@gmail.com wrote:
I have forget to write for outcall in
extension
server1:
[calltoserver2]
exten = _X.,1,Noop(Call to server2)
exten =
_X.,2,Dial(SIP/interboxserver2/${EXTEN})
exten = _X.,3,Hangup
server2:
[calltoserver1]
exten = _X.,1,Noop(Call to server1)
exten =
_X.,2,Dial(SIP/interboxserver1/${EXTEN})
exten = _X.,3,Hangup
:)
Vardan
Vardan wrote:
Hello
Server1:
sip.conf
[interboxserver2]
type=friend
host=192.168.250.112
context=callfromserver2
disallow=all
allow=ulaw
allow=alaw
allow=g729
extensions.conf
[callfromserver2]
exten = _X.,1,Noop(Call from server2)
exten = _X.,2,Dial(SIP/${EXTEN})
exten = _X.,3,Hangup
Server2:
sip.conf
[interboxserver1]
type=friend
host=192.168.250.111
context=callfromserver1
disallow=all
allow=ulaw
allow=alaw
allow=g729
extensions.conf
[callfromserver1]
exten = _X.,1,Noop(Call from server1)
exten = _X.,2,Dial(SIP/${EXTEN})
exten = _X.,3,Hangup
Try so, I think it must work.
And also, look and delete any another records in both
servers in
sip.conf about this servers settings.
Vardan
Vieri wrote:
Hi,
I'm trying to setup a SIP trunk between 2 Asterisk
servers on the same LAN (no NAT, no firewalls).
With IAX2 all's fine but I'm unable to setup SIP.
I must be missing something obvious.
I followed the simple example at
http://www.panoramisk.com/90/sip-trunk-with-asterisk/en/.
so Asterisk server 1 (192.168.250.111) sip.conf
contains:
[interboxsip]
type=peer
host=192.168.250.112
context=mycontext
Asterisk server 2 (192.168.250.112) sip.conf
contains:
[interboxsip]
type=peer
host=192.168.250.111
context=mycontext
I dialed from a SIP extension (4053) in server 2
(192.168.250.112) to 3666 in server 1 (192.168.250.111) via
the interboxsip SIP trunk.
The call fails and according to the SIP messages
it seems to be an authentication problem.
What am I missing?
SIP messages on 192.168.250.112 (Asterisk server 2
- transmitting call):
-- Executing
[3...@from-internal:2] Dial(SIP/4053-6dea,
SIP/interboxsip/3666|300|rt) in new stack
Audio is at 192.168.250.112 port 15850
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to
192.168.250.111:5060:
INVITE sip:3...@192.168.250.111 SIP/2.0
Via: SIP/2.0/UDP
192.168.250.112:5060;branch=z9hG4bK3c951a1d;rport
From:
devicesip:4...@192.168.250.112;tag=as4d17a185
To:sip:3...@192.168.250.111
Contact:sip:4...@192.168.250.112
Call-ID:
3770a8004ce882dd3c89c1d91b5aa...@192.168.250.112
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 12 May 2010 09:13:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 270
v=0
o=root 20611 20611 IN IP4 192.168.250.112
s=session
c=IN IP4 192.168.250.112
t=0 0
m=audio 15850 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called
interboxsip/3666
--- SIP read from 192.168.250.111:5060
---
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.168.250.112:5060;branch=z9hG4bK3c951a1d;received=192.168.250.112;rport=5060
From:
devicesip:4...@192.168.250.112;tag=as4d17a185
To:sip:3...@192.168.250.111;tag=as00842b82
Call-ID:
3770a8004ce882dd3c89c1d91b5aa...@192.168.250.112
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5,
realm=asterisk, nonce=2545a5dd
Content-Length: 0
-
--- (10 headers 0 lines) ---
Transmitting (no NAT) to 192.168.250.111:5060:
ACK sip:3...@192.168.250.111 SIP/2.0
Via: SIP/2.0/UDP
192.168.250.112:5060;branch=z9hG4bK3c951a1d;rport
From:
devicesip:4...@192.168.250.112;tag=as4d17a185
To:sip:3...@192.168.250.111;tag=as00842b82
Contact:sip:4...@192.168.250.112
Call-ID:
3770a8004ce882dd3c89c1d91b5aa...@192.168.250.112
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
--
SIP/interboxsip-6deb is circuit-busy
SIP messages on 192.168.250.111 (Asterisk server 1
- receiving end):
-- SIP read from 192.168.250.112:5060:
INVITE sip:3...@192.168.250.111 SIP/2.0
Via: SIP/2.0/UDP
192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport
From:
devicesip:4...@192.168.250.112;tag=as18a568d6
To:sip:3...@192.168.250.111
Contact:sip:4...@192.168.250.112
Call-ID:
328617546726e5d430538e8061771...@192.168.250.112
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 12 May 2010 09:20:26 GMT
Re: [asterisk-users] Asterisk core dumping on SendFax with FFA
Kevin P. Fleming wrote: Like I said, it's a known problem, and the fix should be out within a day or two. It was reported to us about a week ago, so if you had contacted the support department, it's likely they would have been able to shortcut your hair-pulling experience :-) Hi Kevin, Thanks for the update. Unfortunately I contacted the support team early on in the process (about 35 hours ago) and to date the only response has been Please run the debug process and send us the logs - so there has been much hair pulling in the meanwhile. I have ticket WAJ-201081 logged and am awaiting a response - however it would be appreciated if I could get my hands on the 1.2.1 version and would be more than happy to test it and see if the issue is fixed. Cheers, Ben -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunk between two Asterisk servers
--- On Wed, 5/12/10, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: --- SIP read from 192.168.250.111:5060 --- SIP/2.0 407 Proxy Authentication Required You need to run the SIP debug on 192.168.250.111 to learn more about WHY the 407 is issued. Have a close look and you are likely to understand it right away. Also: Do not forget the reload after applying changes to sip.conf. I always do a sip reload after changes to sip settings. Here are the SIP messages on 192.168.250.111 (Asterisk server 1 - receiving end): -- SIP read from 192.168.250.112:5060: INVITE sip:3...@192.168.250.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport From: device sip:4...@192.168.250.112;tag=as18a568d6 To: sip:3...@192.168.250.111 Contact: sip:4...@192.168.250.112 Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 12 May 2010 09:20:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO upported: replaces Content-Type: application/sdp Content-Length: 270 v=0 o=root 20611 20611 IN IP4 192.168.250.112 s=session c=IN IP4 192.168.250.112 t=0 0 m=audio 14648 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- (14 headers 13 lines) --- Using INVITE request as basis request - 328617546726e5d430538e8061771...@192.168.250.112 Sending to 192.168.250.112 : 5060 (NAT) Reliably Transmitting (NAT) to 192.168.250.112:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;received=192.168.250.112;rport=5060 From: device sip:4...@192.168.250.112;tag=as18a568d6 To: sip:3...@192.168.250.111;tag=as57a19dac Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=1327c5b6 Content-Length: 0 --- Scheduling destruction of call '328617546726e5d430538e8061771...@192.168.250.112' in 15000 ms Found user '4053' -- SIP read from 192.168.250.112:5060: ACK sip:3...@192.168.250.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport From: device sip:4...@192.168.250.112;tag=as18a568d6 To: sip:3...@192.168.250.111;tag=as57a19dac Contact: sip:4...@192.168.250.112 Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 Can you deduce from this what I'm doing wrong? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Stress Test new system
All: Getting ready to put the system in production. Any suggestions on stress testing the system? I'd like to initiate say 10 sip phone calls to make sure the provider has the bandwidth. Can you do that in CLI? I've called 4 numbers simultaneously with the hard phones I currently have and am thinking of adding 6 or so soft-phones to various pc's to make a total of ten outgoing calls at the same time. Any thing else that can be tested before we go live (total of 60 users)? Thanks, Eddie Mikell -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunk between two Asterisk servers
--- On Wed, 5/12/10, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: SIP/2.0 407 Proxy Authentication Required Then you have another entry in sip.conf that uses the same IP address. Delete that, or change the port on one of them, and adjust insecure= accordingly. asterisk1 # grep 192.168.250 sip*.conf sip.conf:host=192.168.250.112 asterisk2 # grep 192.168.250 sip*.conf sip.conf:host=192.168.250.111 So I only have 1 entry in each server's sip.conf and this entry is in interboxsip (my sample SIP trunk name). Puzzling... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunk between two Asterisk servers
Please look in any conf file that have any relations with sip.conf.
I think you have some records.
And one also, you take this message when calling in both direction?
(server1 call server2 and server2 call server1)
Vardan
Vieri wrote:
--- On Wed, 5/12/10, Vardanhvarda...@gmail.com wrote:
I have forget to write for outcall in
extension
server1:
[calltoserver2]
exten = _X.,1,Noop(Call to server2)
exten =
_X.,2,Dial(SIP/interboxserver2/${EXTEN})
exten = _X.,3,Hangup
server2:
[calltoserver1]
exten = _X.,1,Noop(Call to server1)
exten =
_X.,2,Dial(SIP/interboxserver1/${EXTEN})
exten = _X.,3,Hangup
:)
Vardan
Vardan wrote:
Hello
Server1:
sip.conf
[interboxserver2]
type=friend
host=192.168.250.112
context=callfromserver2
disallow=all
allow=ulaw
allow=alaw
allow=g729
extensions.conf
[callfromserver2]
exten = _X.,1,Noop(Call from server2)
exten = _X.,2,Dial(SIP/${EXTEN})
exten = _X.,3,Hangup
Server2:
sip.conf
[interboxserver1]
type=friend
host=192.168.250.111
context=callfromserver1
disallow=all
allow=ulaw
allow=alaw
allow=g729
extensions.conf
[callfromserver1]
exten = _X.,1,Noop(Call from server1)
exten = _X.,2,Dial(SIP/${EXTEN})
exten = _X.,3,Hangup
Try so, I think it must work.
And also, look and delete any another records in both
servers in
sip.conf about this servers settings.
Vardan
Vieri wrote:
Hi,
I'm trying to setup a SIP trunk between 2 Asterisk
servers on the same LAN (no NAT, no firewalls).
With IAX2 all's fine but I'm unable to setup SIP.
I must be missing something obvious.
I followed the simple example at
http://www.panoramisk.com/90/sip-trunk-with-asterisk/en/.
so Asterisk server 1 (192.168.250.111) sip.conf
contains:
[interboxsip]
type=peer
host=192.168.250.112
context=mycontext
Asterisk server 2 (192.168.250.112) sip.conf
contains:
[interboxsip]
type=peer
host=192.168.250.111
context=mycontext
I dialed from a SIP extension (4053) in server 2
(192.168.250.112) to 3666 in server 1 (192.168.250.111) via
the interboxsip SIP trunk.
The call fails and according to the SIP messages
it seems to be an authentication problem.
What am I missing?
SIP messages on 192.168.250.112 (Asterisk server 2
- transmitting call):
-- Executing
[3...@from-internal:2] Dial(SIP/4053-6dea,
SIP/interboxsip/3666|300|rt) in new stack
Audio is at 192.168.250.112 port 15850
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to
192.168.250.111:5060:
INVITE sip:3...@192.168.250.111 SIP/2.0
Via: SIP/2.0/UDP
192.168.250.112:5060;branch=z9hG4bK3c951a1d;rport
From:
devicesip:4...@192.168.250.112;tag=as4d17a185
To:sip:3...@192.168.250.111
Contact:sip:4...@192.168.250.112
Call-ID:
3770a8004ce882dd3c89c1d91b5aa...@192.168.250.112
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 12 May 2010 09:13:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 270
v=0
o=root 20611 20611 IN IP4 192.168.250.112
s=session
c=IN IP4 192.168.250.112
t=0 0
m=audio 15850 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called
interboxsip/3666
--- SIP read from 192.168.250.111:5060
---
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.168.250.112:5060;branch=z9hG4bK3c951a1d;received=192.168.250.112;rport=5060
From:
devicesip:4...@192.168.250.112;tag=as4d17a185
To:sip:3...@192.168.250.111;tag=as00842b82
Call-ID:
3770a8004ce882dd3c89c1d91b5aa...@192.168.250.112
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5,
realm=asterisk, nonce=2545a5dd
Content-Length: 0
-
--- (10 headers 0 lines) ---
Transmitting (no NAT) to 192.168.250.111:5060:
ACK sip:3...@192.168.250.111 SIP/2.0
Via: SIP/2.0/UDP
192.168.250.112:5060;branch=z9hG4bK3c951a1d;rport
From:
devicesip:4...@192.168.250.112;tag=as4d17a185
To:sip:3...@192.168.250.111;tag=as00842b82
Contact:sip:4...@192.168.250.112
Call-ID:
3770a8004ce882dd3c89c1d91b5aa...@192.168.250.112
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
--
SIP/interboxsip-6deb is circuit-busy
SIP messages on 192.168.250.111 (Asterisk server 1
- receiving end):
-- SIP read from 192.168.250.112:5060:
INVITE sip:3...@192.168.250.111 SIP/2.0
Via: SIP/2.0/UDP
192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport
From:
devicesip:4...@192.168.250.112;tag=as18a568d6
To:sip:3...@192.168.250.111
Contact:sip:4...@192.168.250.112
Call-ID:
328617546726e5d430538e8061771...@192.168.250.112
CSeq: 102 INVITE
Re: [asterisk-users] SIP trunk between two Asterisk servers
please show sip show users and sip show peers vardan Vieri wrote: --- On Wed, 5/12/10, Philipp von Klitzingklitz...@pool.informatik.rwth-aachen.de wrote: --- SIP read from 192.168.250.111:5060 --- SIP/2.0 407 Proxy Authentication Required You need to run the SIP debug on 192.168.250.111 to learn more about WHY the 407 is issued. Have a close look and you are likely to understand it right away. Also: Do not forget the reload after applying changes to sip.conf. I always do a sip reload after changes to sip settings. Here are the SIP messages on 192.168.250.111 (Asterisk server 1 - receiving end): -- SIP read from 192.168.250.112:5060: INVITE sip:3...@192.168.250.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport From: devicesip:4...@192.168.250.112;tag=as18a568d6 To:sip:3...@192.168.250.111 Contact:sip:4...@192.168.250.112 Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 12 May 2010 09:20:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO upported: replaces Content-Type: application/sdp Content-Length: 270 v=0 o=root 20611 20611 IN IP4 192.168.250.112 s=session c=IN IP4 192.168.250.112 t=0 0 m=audio 14648 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- (14 headers 13 lines) --- Using INVITE request as basis request - 328617546726e5d430538e8061771...@192.168.250.112 Sending to 192.168.250.112 : 5060 (NAT) Reliably Transmitting (NAT) to 192.168.250.112:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;received=192.168.250.112;rport=5060 From: devicesip:4...@192.168.250.112;tag=as18a568d6 To:sip:3...@192.168.250.111;tag=as57a19dac Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=1327c5b6 Content-Length: 0 --- Scheduling destruction of call '328617546726e5d430538e8061771...@192.168.250.112' in 15000 ms Found user '4053' -- SIP read from 192.168.250.112:5060: ACK sip:3...@192.168.250.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport From: devicesip:4...@192.168.250.112;tag=as18a568d6 To:sip:3...@192.168.250.111;tag=as57a19dac Contact:sip:4...@192.168.250.112 Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 Can you deduce from this what I'm doing wrong? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 70, Issue 25
Hi Vardan
I did same as you told and deleted the SIP information in Astdb and
restarted asterisk. but the result was same.
as you said there might be mistake in sip.conf so i am pasting both servers
configuration here..
1- nasir.server.com
[abc]
username=abc
type=friend
secret=mysecret
nat=yes
mailbox=12234568
incominglimit=2
outgoinglimit=2
host=dynamic
dtmfmode=rfc2833
context=payasyougo
canreinvite=yes
callerid=Nasir Qazi 12234
accountcode=6:0:abc
amaflags=default
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm
2- 192.168.0.254 (client system)
[abc]
type=peer
username=abc
secret=mysecret
host=nasir.server.com
context=default
dtmfmode=rfc2833
canreinvite=yes
insecure=very
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm
nat=yes
;qualify=yes
[caller]
type=friend
secret=123456
host=dynamic
callerid=caller 1212988
context=out
nat=yes
dtmfmode=rfc2833
canreinvite=yes
insecure=no
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm
t38_udptl=yes
qualify=yes
I have registered [caller] on xlite at client system and dialing following
context in local system that will dial [abc]
[out]
exten= _X.,1,Dial(SIP/${ext...@abc,30,1)
exten= _X.,n,Hangup
as you can see above *highlighted that context of abc is
payasyougo.*problem is that i want the call to land in that context on
nasir.server.com, which works if i use register string. but without register
string call goes to default context on nasir.server.com
regards,
Nasir Javaid
Message: 19
Date: Tue, 11 May 2010 20:54:30 +0500
From: Vardan hvarda...@gmail.com
Subject: Re: [asterisk-users] asterisk-users Digest, Vol 70, Issue 24
To: asterisk-users@lists.digium.com
Message-ID: hsbujk$qk...@dough.gmane.org
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hello Nasir
I have some please.
Do so, it help.
Find all records about interexchange beetwen this two server and delete
all records in sip.conf for this both server (first make backup
sip.conf, or any another conf file that you use).
restart asterisk.
look in astdb about this old records, if any found, delete him
Next, create new record in sip.conf on both servers, without
registration string, reload sip.conf.
give him right context from extensions.conf.
Can you do this?
I think is some mistake about configuration in sip.conf, you have I
think two same records (peer or friend).
Vardan
--
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Re: [asterisk-users] Stress Test new system
Here's one way - set up calls to the sip provider using local channels instead of actual phones. In extensions.conf [monkeys] Exten = s,1,playback(tt-monkeys) Exten = s,n,hangup Create Call file (monkey1.call) Channel: sip/5551212 CallerID: Local/8 MaxRetries: 1 WaitTime: 60 retryTime: 5 extension: local/8 Context: monkeys Cp monkey1.call /var/spool/asterisk/outgoing Copy monkey1.call to monkeyx.call for each line you want to test I use this methodology to test my asterisk for 25 simultaneous calls. If you change local/8 to sip/123 where 123 is one of your physical extensions, this places a real call. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eddie Mikell Sent: Wednesday, May 12, 2010 9:00 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Stress Test new system All: Getting ready to put the system in production. Any suggestions on stress testing the system? I'd like to initiate say 10 sip phone calls to make sure the provider has the bandwidth. Can you do that in CLI? I've called 4 numbers simultaneously with the hard phones I currently have and am thinking of adding 6 or so soft-phones to various pc's to make a total of ten outgoing calls at the same time. Any thing else that can be tested before we go live (total of 60 users)? Thanks, Eddie Mikell -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunk between two Asterisk servers
And sip show registry Vardan Vieri wrote: --- On Wed, 5/12/10, Philipp von Klitzingklitz...@pool.informatik.rwth-aachen.de wrote: --- SIP read from 192.168.250.111:5060 --- SIP/2.0 407 Proxy Authentication Required You need to run the SIP debug on 192.168.250.111 to learn more about WHY the 407 is issued. Have a close look and you are likely to understand it right away. Also: Do not forget the reload after applying changes to sip.conf. I always do a sip reload after changes to sip settings. Here are the SIP messages on 192.168.250.111 (Asterisk server 1 - receiving end): -- SIP read from 192.168.250.112:5060: INVITE sip:3...@192.168.250.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport From: devicesip:4...@192.168.250.112;tag=as18a568d6 To:sip:3...@192.168.250.111 Contact:sip:4...@192.168.250.112 Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 12 May 2010 09:20:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO upported: replaces Content-Type: application/sdp Content-Length: 270 v=0 o=root 20611 20611 IN IP4 192.168.250.112 s=session c=IN IP4 192.168.250.112 t=0 0 m=audio 14648 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- (14 headers 13 lines) --- Using INVITE request as basis request - 328617546726e5d430538e8061771...@192.168.250.112 Sending to 192.168.250.112 : 5060 (NAT) Reliably Transmitting (NAT) to 192.168.250.112:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;received=192.168.250.112;rport=5060 From: devicesip:4...@192.168.250.112;tag=as18a568d6 To:sip:3...@192.168.250.111;tag=as57a19dac Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=1327c5b6 Content-Length: 0 --- Scheduling destruction of call '328617546726e5d430538e8061771...@192.168.250.112' in 15000 ms Found user '4053' -- SIP read from 192.168.250.112:5060: ACK sip:3...@192.168.250.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport From: devicesip:4...@192.168.250.112;tag=as18a568d6 To:sip:3...@192.168.250.111;tag=as57a19dac Contact:sip:4...@192.168.250.112 Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 Can you deduce from this what I'm doing wrong? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 70, Issue 25
here i am attaching debug trace of sip in case of sccessfull call when using register string... *CLI [May 12 19:21:14] --- SIP read from 192.168.0.254:5060 --- INVITE sip:17185594...@nasir.server.com sip%3a17185594...@nasir.server.comSIP/2.0 Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK3c63f272;rport Max-Forwards: 70 From: caller sip:12129887...@192.168.0.254sip%3a12129887...@192.168.0.254 ;tag=as76623e31 To: sip:17185594...@nasir.server.com sip%3a17185594...@nasir.server.com Contact: sip:12129887...@192.168.0.254 sip%3a12129887...@192.168.0.254 Call-ID: 245c407103141a6841c0ac106bd5a...@192.168.0.254 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.0 Date: Wed, 12 May 2010 14:20:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 284 v=0 o=root 618893758 618893758 IN IP4 192.168.0.254 s=Asterisk PBX 1.6.2.0 c=IN IP4 192.168.0.254 t=0 0 m=audio 11026 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv - [May 12 19:21:14] --- (14 headers 13 lines) --- [May 12 19:21:14] Sending to 192.168.0.254 : 5060 (NAT) [May 12 19:21:14] Using INVITE request as basis request - 245c407103141a6841c0ac106bd5a...@192.168.0.254 [May 12 19:21:14] Found peer 'abc' [May 12 19:21:14] --- Reliably Transmitting (NAT) to 192.168.0.254:5060 --- SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.254:5060 ;branch=z9hG4bK3c63f272;received=192.168.0.254;rport=5060 From: caller sip:12129887...@192.168.0.254sip%3a12129887...@192.168.0.254 ;tag=as76623e31 To: sip:17185594...@nasir.server.com sip%3a17185594...@nasir.server.com ;tag=as0a721b3a Call-ID: 245c407103141a6841c0ac106bd5a...@192.168.0.254 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=7bc52d0a Content-Length: 0 [May 12 19:21:14] Scheduling destruction of SIP dialog ' 245c407103141a6841c0ac106bd5a...@192.168.0.254' in 32000 ms (Method: INVITE) [May 12 19:21:14] --- SIP read from 192.168.0.254:5060 --- ACK sip:17185594...@nasir.server.com sip%3a17185594...@nasir.server.comSIP/2.0 Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK3c63f272;rport Max-Forwards: 70 From: caller sip:12129887...@192.168.0.254sip%3a12129887...@192.168.0.254 ;tag=as76623e31 To: sip:17185594...@nasir.server.com sip%3a17185594...@nasir.server.com ;tag=as0a721b3a Contact: sip:12129887...@192.168.0.254 sip%3a12129887...@192.168.0.254 Call-ID: 245c407103141a6841c0ac106bd5a...@192.168.0.254 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.2.0 Content-Length: 0 - [May 12 19:21:14] --- (10 headers 0 lines) --- [May 12 19:21:14] --- SIP read from 192.168.0.254:5060 --- INVITE sip:17185594...@nasir.server.com sip%3a17185594...@nasir.server.comSIP/2.0 Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK05611806;rport Max-Forwards: 70 From: caller sip:12129887...@192.168.0.254sip%3a12129887...@192.168.0.254 ;tag=as76623e31 To: sip:17185594...@nasir.server.com sip%3a17185594...@nasir.server.com Contact: sip:12129887...@192.168.0.254 sip%3a12129887...@192.168.0.254 Call-ID: 245c407103141a6841c0ac106bd5a...@192.168.0.254 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.6.2.0 Proxy-Authorization: Digest username=abc, realm=asterisk, algorithm=MD5, uri=sip:17185594...@nasir.server.com sip%3a17185594...@nasir.server.com, nonce=7bc52d0a, response=f138ecd92bb706207a7b8d00f1c1bed7 Date: Wed, 12 May 2010 14:20:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 284 v=0 o=root 618893758 618893759 IN IP4 192.168.0.254 s=Asterisk PBX 1.6.2.0 c=IN IP4 192.168.0.254 t=0 0 m=audio 11026 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv - [May 12 19:21:14] --- (15 headers 13 lines) --- [May 12 19:21:14] Sending to 192.168.0.254 : 5060 (NAT) [May 12 19:21:14] Using INVITE request as basis request - 245c407103141a6841c0ac106bd5a...@192.168.0.254 [May 12 19:21:14] Found peer 'abc' [May 12 19:21:14] Found RTP audio format 0 [May 12 19:21:14] Found RTP audio format 3 [May 12 19:21:14] Found RTP audio format 101 [May 12 19:21:14] Peer audio RTP is at port 192.168.0.254:11026 [May 12 19:21:14] Found description format PCMU for ID 0 [May 12 19:21:14] Found description format GSM for ID 3 [May 12 19:21:14] Found description format telephone-event for ID 101 [May 12 19:21:14] Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x6 (gsm|ulaw)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw) [May 12 19:21:14] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Re: [asterisk-users] Stress Test new system
If you can call yourself via the provider just setup a dialplan which spirals the call,e.g. from softphone call via provider one of your numbers. Then incoming call route to your next DID, and so on, and after some spiraling just connect the call to the Milliwatt() application. Milliwatt is perfect to hear packet loss and jitter. regards klaus Am 12.05.2010 15:59, schrieb Eddie Mikell: All: Getting ready to put the system in production. Any suggestions on stress testing the system? I'd like to initiate say 10 sip phone calls to make sure the provider has the bandwidth. Can you do that in CLI? I've called 4 numbers simultaneously with the hard phones I currently have and am thinking of adding 6 or so soft-phones to various pc's to make a total of ten outgoing calls at the same time. Any thing else that can be tested before we go live (total of 60 users)? Thanks, Eddie Mikell -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 70, Issue 25
Look, you do again with registration.
remove any registration information.
Look this config, I think it can help you
Server1:
sip.conf
[interboxserver2]
type=friend
host=192.168.250.112
context=callfromserver2
disallow=all
allow=ulaw
allow=alaw
allow=g729
extensions.conf
[calltoserver2]
exten = _X.,1,Noop(Call to server2)
exten = _X.,2,Dial(SIP/interboxserver2/${EXTEN})
exten = _X.,3,Hangup
[callfromserver2]
exten = _X.,1,Noop(Call from server2)
exten = _X.,2,Dial(SIP/${EXTEN})
exten = _X.,3,Hangup
Server2:
sip.conf
[interboxserver1]
type=friend
host=192.168.250.111
context=callfromserver1
disallow=all
allow=ulaw
allow=alaw
allow=g729
extensions.conf
[calltoserver1]
exten = _X.,1,Noop(Call to server1)
exten = _X.,2,Dial(SIP/interboxserver1/${EXTEN})
exten = _X.,3,Hangup
[callfromserver1]
exten = _X.,1,Noop(Call from server1)
exten = _X.,2,Dial(SIP/${EXTEN})
exten = _X.,3,Hangup
Try so, I think it must work.
And also, look and delete any another records in both servers in
sip.conf about this servers settings.
Vardan
Nasir Javaid wrote:
Hi Vardan
I did same as you told and deleted the SIP information in Astdb and
restarted asterisk. but the result was same.
as you said there might be mistake in sip.conf so i am pasting both
servers configuration here..
1- nasir.server.com http://nasir.server.com
[abc]
username=abc
type=friend
secret=mysecret
nat=yes
mailbox=12234568
incominglimit=2
outgoinglimit=2
host=dynamic
dtmfmode=rfc2833
context=payasyougo
canreinvite=yes
callerid=Nasir Qazi 12234
accountcode=6:0:abc
amaflags=default
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm
2- 192.168.0.254 (client system)
[abc]
type=peer
username=abc
secret=mysecret
host=nasir.server.com http://nasir.server.com
context=default
dtmfmode=rfc2833
canreinvite=yes
insecure=very
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm
nat=yes
;qualify=yes
[caller]
type=friend
secret=123456
host=dynamic
callerid=caller 1212988
context=out
nat=yes
dtmfmode=rfc2833
canreinvite=yes
insecure=no
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm
t38_udptl=yes
qualify=yes
I have registered [caller] on xlite at client system and dialing
following context in local system that will dial [abc]
[out]
exten= _X.,1,Dial(SIP/${ext...@abc,30,1)
exten= _X.,n,Hangup
as you can see above *highlighted that context of abc is payasyougo.*
problem is that i want the call to land in that context on
nasir.server.com http://nasir.server.com, which works if i use
register string. but without register string call goes to default
context on nasir.server.com http://nasir.server.com
regards,
Nasir Javaid
Message: 19
Date: Tue, 11 May 2010 20:54:30 +0500
From: Vardan hvarda...@gmail.com mailto:hvarda...@gmail.com
Subject: Re: [asterisk-users] asterisk-users Digest, Vol 70, Issue 24
To: asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com
Message-ID: hsbujk$qk...@dough.gmane.org mailto:1...@dough.gmane.org
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hello Nasir
I have some please.
Do so, it help.
Find all records about interexchange beetwen this two server and delete
all records in sip.conf for this both server (first make backup
sip.conf, or any another conf file that you use).
restart asterisk.
look in astdb about this old records, if any found, delete him
Next, create new record in sip.conf on both servers, without
registration string, reload sip.conf.
give him right context from extensions.conf.
Can you do this?
I think is some mistake about configuration in sip.conf, you have I
think two same records (peer or friend).
Vardan
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Re: [asterisk-users] Asterisk Bible?
Regarding functions and applications options, the only authoritative source is the console: core show application ... core show function ... regards Klaus Am 07.05.2010 18:37, schrieb Tim Densmore: Hi Folks, Is there a generally accepted Asterisk bible for current versions? I poked around the forums and there didn't seem to be a real consensus, and there are lots of options out there. I need something that focuses on Asterisk dialplans and config files, not a linux primer. I'm looking for dead-tree rather than online documentation. Thanks, Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 70, Issue 25
Hi again, below is debug trace of * cli when i remove register string from sip.conf *CLI [May 12 19:33:06] --- SIP read from 192.168.0.254:5060 --- INVITE sip:17185594...@nasir.server.com sip%3a17185594...@nasir.server.comSIP/2.0 Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK56e3b44a;rport Max-Forwards: 70 From: caller sip:12129887...@192.168.0.254sip%3a12129887...@192.168.0.254 ;tag=as5b6db7a2 To: sip:17185594...@nasir.server.com sip%3a17185594...@nasir.server.com Contact: sip:12129887...@192.168.0.254 sip%3a12129887...@192.168.0.254 Call-ID: 23c4c49b329104d31ad6822c02cb8...@192.168.0.254 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.0 Date: Wed, 12 May 2010 14:32:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 284 v=0 o=root 814806874 814806874 IN IP4 192.168.0.254 s=Asterisk PBX 1.6.2.0 c=IN IP4 192.168.0.254 t=0 0 m=audio 17632 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv - [May 12 19:33:06] --- (14 headers 13 lines) --- [May 12 19:33:06] Sending to 192.168.0.254 : 5060 (NAT) [May 12 19:33:06] Using INVITE request as basis request - 23c4c49b329104d31ad6822c02cb8...@192.168.0.254 [May 12 19:33:06] Found no matching peer or user for '192.168.0.254:5060' [May 12 19:33:06] Found RTP audio format 0 [May 12 19:33:06] Found RTP audio format 3 [May 12 19:33:06] Found RTP audio format 101 [May 12 19:33:06] Peer audio RTP is at port 192.168.0.254:17632 [May 12 19:33:06] Found description format PCMU for ID 0 [May 12 19:33:06] Found description format GSM for ID 3 [May 12 19:33:06] Found description format telephone-event for ID 101 [May 12 19:33:06] Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x6 (gsm|ulaw)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw) [May 12 19:33:06] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [May 12 19:33:06] Peer audio RTP is at port 192.168.0.254:17632 [May 12 19:33:06] Looking for 17185594743 in default (domain nasir.server.com) [May 12 19:33:06] WARNING[4113]: chan_sip.c:3930 sip_new: setting callerid number to 1212988 [May 12 19:33:06] list_route: hop: sip:12129887...@192.168.0.254sip%3a12129887...@192.168.0.254 [May 12 19:33:06] --- Transmitting (NAT) to 192.168.0.254:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.254:5060 ;branch=z9hG4bK56e3b44a;received=192.168.0.254;rport=5060 From: caller sip:12129887...@192.168.0.254sip%3a12129887...@192.168.0.254 ;tag=as5b6db7a2 To: sip:17185594...@nasir.server.com sip%3a17185594...@nasir.server.com Call-ID: 23c4c49b329104d31ad6822c02cb8...@192.168.0.254 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:17185594...@nasir.server.comsip%3a17185594...@nasir.server.com Content-Length: 0 On Wed, May 12, 2010 at 7:26 PM, Nasir Javaid nasirjavaidna...@gmail.comwrote: here i am attaching debug trace of sip in case of sccessfull call when using register string... *CLI [May 12 19:21:14] --- SIP read from 192.168.0.254:5060 --- INVITE sip:17185594...@nasir.server.comsip%3a17185594...@nasir.server.comSIP/2.0 Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK3c63f272;rport Max-Forwards: 70 From: caller sip:12129887...@192.168.0.254sip%3a12129887...@192.168.0.254 ;tag=as76623e31 To: sip:17185594...@nasir.server.com sip%3a17185594...@nasir.server.com Contact: sip:12129887...@192.168.0.254 sip%3a12129887...@192.168.0.254 Call-ID: 245c407103141a6841c0ac106bd5a...@192.168.0.254 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.0 Date: Wed, 12 May 2010 14:20:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 284 v=0 o=root 618893758 618893758 IN IP4 192.168.0.254 s=Asterisk PBX 1.6.2.0 c=IN IP4 192.168.0.254 t=0 0 m=audio 11026 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv - [May 12 19:21:14] --- (14 headers 13 lines) --- [May 12 19:21:14] Sending to 192.168.0.254 : 5060 (NAT) [May 12 19:21:14] Using INVITE request as basis request - 245c407103141a6841c0ac106bd5a...@192.168.0.254 [May 12 19:21:14] Found peer 'abc' [May 12 19:21:14] --- Reliably Transmitting (NAT) to 192.168.0.254:5060 --- SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.254:5060 ;branch=z9hG4bK3c63f272;received=192.168.0.254;rport=5060 From: caller sip:12129887...@192.168.0.254sip%3a12129887...@192.168.0.254 ;tag=as76623e31 To: sip:17185594...@nasir.server.com sip%3a17185594...@nasir.server.com ;tag=as0a721b3a Call-ID: 245c407103141a6841c0ac106bd5a...@192.168.0.254
Re: [asterisk-users] Possible bug in chan_sip:add_sdp
This code is really ugly und hard to verify. Please file a bug report at https://issues.asterisk.org/ thanks klaus Am 06.05.2010 23:54, schrieb Richard Kenner: I can confirm that the following fixes my problem: --- chan_sip.c (revision 261450) +++ chan_sip.c (working copy) @@ -10357,12 +10357,22 @@ strlen(connection) + strlen(session_time); if (needaudio) len += m_audio-used + a_audio-used + strlen(hold); + else if (p-offered_media[SDP_AUDIO].offered) + len += strlen(m=audio 0 RTP/AVP \r\n) + strlen(p-offered_media[SDP_AUDIO].text); + if (needvideo) /* only if video response is appropriate */ len += m_video-used + a_video-used + strlen(bandwidth) + strlen(hold); + else if (p-offered_media[SDP_VIDEO].offered) + len += strlen(m=video 0 RTP/AVP \r\n) + strlen(p-offered_media[SDP_VIDEO].text); + if (needtext) /* only if text response is appropriate */ len += m_text-used + a_text-used + strlen(hold); + else if (p-offered_media[SDP_TEXT].offered) + len += strlen(m=text 0 RTP/AVP \r\n) + strlen(p-offered_media[SDP_TEXT].text); if (add_t38) len += m_modem-used + a_modem-used; + else if (p-offered_media[SDP_IMAGE].offered) + len += strlen(m=image 0 udptl t38\r\n); add_header(resp, Content-Type, application/sdp); add_header_contentLength(resp, len); -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] meetme and jitterbuffer
hi all. When I use conference call, my setting about jitterbuffer on sip.conf doesn't work. ### sip.conf # jbenable = yes jbforce = yes jbmaxsize = 100 jbresyncthreshold = 1000 jbimpl = fixed ### And I understood how to be effective jitterbuffer on conference call. I have to use nj option in extention. Is this right ? Because of non-effective setting on sip.conf, I can't change length of the jitterbuffer. When I use conference call, the delay of conversation is about 210 ms. It's too big delay. In fixedjitterbuffer.h , /* defaults */ #define FIXED_JB_SIZE_DEFAULT 200 #define FIXED_JB_RESYNCH_THRESHOLD_DEFAULT 1000 This FIXED_JB_SIZE_DEFAULT 200 causes 210 ms delay. If I change this to 100, the delay was about 110 on conference call. Finally if I want to change the length of jitterbuffer on conference call, I have to change source code every time. It's only way. Is this right ? I think this is too inconvenient. thx. - nakaji -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Bible?
Un-top-posting... Am 07.05.2010 18:37, schrieb Tim Densmore: Is there a generally accepted Asterisk bible for current versions? I poked around the forums and there didn't seem to be a real consensus, and there are lots of options out there. I need something that focuses on Asterisk dialplans and config files, not a linux primer. I'm looking for dead-tree rather than online documentation. On Wed, 12 May 2010, Klaus Darilion wrote: Regarding functions and applications options, the only authoritative source is the console: core show application ... core show function ... Assuming you don't use autoload=no in modules.conf. I usually only load about 25 modules. Dead-tree dox will always trail a fast moving project. Current is also in the eye of the beholder. Stable, production ready, or trunk? I'm still using 1.2 on all of my production systems. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Possible bug in chan_sip:add_sdp
This code is really ugly und hard to verify. Since the computation of the is being done with separate code from the actual output, the code in that part of the module is indeed ugly. But I wanted to make the smallest possible change. However, I do suggest that the full output string be built up and the output as once. Please file a bug report at https://issues.asterisk.org/ Will do. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] bad magic number log messages
Anyone else get this issue - around 200 entries per second of this in the Asterisk messages file: astobj2.c:115 INTERNAL_OBJ: bad magic number 0x27b4113a Seems to happen after several hours of receiving a steady stream of test calls. My messages file is 7.5 gigs... John -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need fax solution for 1.4.xx
You are right that PIKA no longer just sells Fax licenses to be used with 3rd party boards. However the PIKA Warp appliance is great for Faxing with Asterisk. http://www.pikatechnologies.com/english/View.asp?x=1009 Rod == On Tue, May 11, 2010 at 3:30 PM, William Stillwell (Lists) william.stillwell-li...@ wrote: Anybody know a reliable fax solution for 1.4.30 branch? I am using PikaFax on another server and works very well (about 3000 faxes a week), but it appears they no longer offer their product to open source asterisk, only for there WARP appliance. NOT really looking to migrate from 1.4.x to 1.6.x -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No ringtone when going from queue to dial-command
Yes, 20 in Queue is timeout... works fine. Also with the Ringing() command, there is no dialtone... It's just silence... With or without the r-option, always the same. When there is no Queue in between the 2 dial-commands, then the ringtone is there as it should be ! So when I change to the Queue and to musiconhold, I loose the ringtone... Should I do something after the Queue-command to get the ringing back ?? Ringing() does not help in my case... Jonas. On 05/12/2010 12:03 PM, Vardan wrote: Try so: 1. dial(SIP/account1,20) 2. queue(myqueue,,,20) 3. Ringing 4. dial(SIP/account2,,r) 20 in queue is timeout? http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial Vardan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need fax solution for 1.4.xx
But that can't handle the call volume, and doesn't support (2) 23B+D now does it? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rod Boileau Sent: Wednesday, May 12, 2010 11:23 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Need fax solution for 1.4.xx You are right that PIKA no longer just sells Fax licenses to be used with 3rd party boards. However the PIKA Warp appliance is great for Faxing with Asterisk. http://www.pikatechnologies.com/english/View.asp?x=1009 Rod == On Tue, May 11, 2010 at 3:30 PM, William Stillwell (Lists) william.stillwell-li...@ wrote: Anybody know a reliable fax solution for 1.4.30 branch? I am using PikaFax on another server and works very well (about 3000 faxes a week), but it appears they no longer offer their product to open source asterisk, only for there WARP appliance. NOT really looking to migrate from 1.4.x to 1.6.x -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunk between two Asterisk servers
--- On Wed, 5/12/10, Vardan hvarda...@gmail.com wrote: please show sip show users and sip show peers SERVER 2: sip show users (trimmed to just my sip test trunk): Username Secret Accountcode Def.Context ACL NAT interboxsip mycontext No RFC3581 sip show peers (also trimmed): Name/username HostDyn Nat ACL Port Status sipprovider/01 w.x.y.zN 5060 OK (90 ms) interboxsip192.168.250.111 5060 Unmonitored 7503/7503 10.215.146.190 D N A 5060 OK (20 ms) 7502/7502 10.215.146.203 D N A 5060 OK (20 ms) 7172/7172 192.168.250.7D N A 13404OK (40 ms) 7166/7166 10.215.146.200 D N A 5060 OK (20 ms) 7165/7165 10.215.248.12D N A 5060 OK (1 ms) 7160/7160 10.215.146.182 D N A 5060 OK (20 ms) 7137/7137 192.168.250.6D N A 25967OK (10 ms) 7118/7118 192.168.250.10 D N A 14508OK (1 ms) 7117/7117 10.215.146.185 D N A 5060 OK (20 ms) 7114/7114 192.168.250.8D N A 12342OK (10 ms) 7112/7112 192.168.250.31 D N A 19829OK (10 ms) 7111/7111 192.168.250.32 D N A 35259OK (80 ms) 7109/7109 (Unspecified)D N A 0UNKNOWN 7097/7097 10.215.146.164 D N A 5060 OK (20 ms) SERVER 1: sip show users is identical. sip show peers (trimmed): Name/username HostDyn Nat ACL Port Status sipprovider/01 w.x.y.zN 5060 OK (79 ms) interboxsip192.168.250.112 5060 Unmonitored vardan Vieri wrote: --- On Wed, 5/12/10, Philipp von Klitzingklitz...@pool.informatik.rwth-aachen.de wrote: --- SIP read from 192.168.250.111:5060 --- SIP/2.0 407 Proxy Authentication Required You need to run the SIP debug on 192.168.250.111 to learn more about WHY the 407 is issued. Have a close look and you are likely to understand it right away. Also: Do not forget the reload after applying changes to sip.conf. I always do a sip reload after changes to sip settings. Here are the SIP messages on 192.168.250.111 (Asterisk server 1 - receiving end): -- SIP read from 192.168.250.112:5060: INVITE sip:3...@192.168.250.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport From: devicesip:4...@192.168.250.112;tag=as18a568d6 To:sip:3...@192.168.250.111 Contact:sip:4...@192.168.250.112 Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 12 May 2010 09:20:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO upported: replaces Content-Type: application/sdp Content-Length: 270 v=0 o=root 20611 20611 IN IP4 192.168.250.112 s=session c=IN IP4 192.168.250.112 t=0 0 m=audio 14648 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- (14 headers 13 lines) --- Using INVITE request as basis request - 328617546726e5d430538e8061771...@192.168.250.112 Sending to 192.168.250.112 : 5060 (NAT) Reliably Transmitting (NAT) to 192.168.250.112:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;received=192.168.250.112;rport=5060 From: devicesip:4...@192.168.250.112;tag=as18a568d6 To:sip:3...@192.168.250.111;tag=as57a19dac Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=1327c5b6 Content-Length: 0 --- Scheduling destruction of call '328617546726e5d430538e8061771...@192.168.250.112' in 15000 ms Found user '4053' -- SIP read from 192.168.250.112:5060: ACK sip:3...@192.168.250.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport From: devicesip:4...@192.168.250.112;tag=as18a568d6 To:sip:3...@192.168.250.111;tag=as57a19dac Contact:sip:4...@192.168.250.112 Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 Can you deduce from this what I'm doing wrong?
Re: [asterisk-users] SIP trunk between two Asterisk servers
--- On Wed, 5/12/10, Vardan hvarda...@gmail.com wrote: And sip show registry sip show registry doesn't list anything regarding my interboxsip test trunk because I'm trying to setup a straightforward link such as this one described here (without user/password): http://www.panoramisk.com/90/sip-trunk-with-asterisk/en/ The only sip show registry entry I have is the one for my external Internet SIP trunk, which is ok. Thanks for your time. Vardan Vieri wrote: --- On Wed, 5/12/10, Philipp von Klitzingklitz...@pool.informatik.rwth-aachen.de wrote: --- SIP read from 192.168.250.111:5060 --- SIP/2.0 407 Proxy Authentication Required You need to run the SIP debug on 192.168.250.111 to learn more about WHY the 407 is issued. Have a close look and you are likely to understand it right away. Also: Do not forget the reload after applying changes to sip.conf. I always do a sip reload after changes to sip settings. Here are the SIP messages on 192.168.250.111 (Asterisk server 1 - receiving end): -- SIP read from 192.168.250.112:5060: INVITE sip:3...@192.168.250.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport From: devicesip:4...@192.168.250.112;tag=as18a568d6 To:sip:3...@192.168.250.111 Contact:sip:4...@192.168.250.112 Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 12 May 2010 09:20:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO upported: replaces Content-Type: application/sdp Content-Length: 270 v=0 o=root 20611 20611 IN IP4 192.168.250.112 s=session c=IN IP4 192.168.250.112 t=0 0 m=audio 14648 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- (14 headers 13 lines) --- Using INVITE request as basis request - 328617546726e5d430538e8061771...@192.168.250.112 Sending to 192.168.250.112 : 5060 (NAT) Reliably Transmitting (NAT) to 192.168.250.112:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;received=192.168.250.112;rport=5060 From: devicesip:4...@192.168.250.112;tag=as18a568d6 To:sip:3...@192.168.250.111;tag=as57a19dac Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=1327c5b6 Content-Length: 0 --- Scheduling destruction of call '328617546726e5d430538e8061771...@192.168.250.112' in 15000 ms Found user '4053' -- SIP read from 192.168.250.112:5060: ACK sip:3...@192.168.250.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport From: devicesip:4...@192.168.250.112;tag=as18a568d6 To:sip:3...@192.168.250.111;tag=as57a19dac Contact:sip:4...@192.168.250.112 Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 Can you deduce from this what I'm doing wrong? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunk between two Asterisk servers
Please change the peers name in any server. for example: server1: interboxsip1 server2: interboxsip2 Vardan Vieri wrote: --- On Wed, 5/12/10, Vardanhvarda...@gmail.com wrote: please show sip show users and sip show peers SERVER 2: sip show users (trimmed to just my sip test trunk): Username Secret Accountcode Def.Context ACL NAT interboxsip mycontext No RFC3581 sip show peers (also trimmed): Name/username HostDyn Nat ACL Port Status sipprovider/01 w.x.y.zN 5060 OK (90 ms) interboxsip192.168.250.111 5060 Unmonitored 7503/7503 10.215.146.190 D N A 5060 OK (20 ms) 7502/7502 10.215.146.203 D N A 5060 OK (20 ms) 7172/7172 192.168.250.7D N A 13404OK (40 ms) 7166/7166 10.215.146.200 D N A 5060 OK (20 ms) 7165/7165 10.215.248.12D N A 5060 OK (1 ms) 7160/7160 10.215.146.182 D N A 5060 OK (20 ms) 7137/7137 192.168.250.6D N A 25967OK (10 ms) 7118/7118 192.168.250.10 D N A 14508OK (1 ms) 7117/7117 10.215.146.185 D N A 5060 OK (20 ms) 7114/7114 192.168.250.8D N A 12342OK (10 ms) 7112/7112 192.168.250.31 D N A 19829OK (10 ms) 7111/7111 192.168.250.32 D N A 35259OK (80 ms) 7109/7109 (Unspecified)D N A 0UNKNOWN 7097/7097 10.215.146.164 D N A 5060 OK (20 ms) SERVER 1: sip show users is identical. sip show peers (trimmed): Name/username HostDyn Nat ACL Port Status sipprovider/01 w.x.y.zN 5060 OK (79 ms) interboxsip192.168.250.112 5060 Unmonitored vardan Vieri wrote: --- On Wed, 5/12/10, Philipp von Klitzingklitz...@pool.informatik.rwth-aachen.de wrote: --- SIP read from 192.168.250.111:5060 --- SIP/2.0 407 Proxy Authentication Required You need to run the SIP debug on 192.168.250.111 to learn more about WHY the 407 is issued. Have a close look and you are likely to understand it right away. Also: Do not forget the reload after applying changes to sip.conf. I always do a sip reload after changes to sip settings. Here are the SIP messages on 192.168.250.111 (Asterisk server 1 - receiving end): -- SIP read from 192.168.250.112:5060: INVITE sip:3...@192.168.250.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport From: devicesip:4...@192.168.250.112;tag=as18a568d6 To:sip:3...@192.168.250.111 Contact:sip:4...@192.168.250.112 Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 12 May 2010 09:20:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO upported: replaces Content-Type: application/sdp Content-Length: 270 v=0 o=root 20611 20611 IN IP4 192.168.250.112 s=session c=IN IP4 192.168.250.112 t=0 0 m=audio 14648 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- (14 headers 13 lines) --- Using INVITE request as basis request - 328617546726e5d430538e8061771...@192.168.250.112 Sending to 192.168.250.112 : 5060 (NAT) Reliably Transmitting (NAT) to 192.168.250.112:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;received=192.168.250.112;rport=5060 From: devicesip:4...@192.168.250.112;tag=as18a568d6 To:sip:3...@192.168.250.111;tag=as57a19dac Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=1327c5b6 Content-Length: 0 --- Scheduling destruction of call '328617546726e5d430538e8061771...@192.168.250.112' in 15000 ms Found user '4053' -- SIP read from 192.168.250.112:5060: ACK sip:3...@192.168.250.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport From: devicesip:4...@192.168.250.112;tag=as18a568d6 To:sip:3...@192.168.250.111;tag=as57a19dac Contact:sip:4...@192.168.250.112 Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 Can you deduce from this what I'm doing wrong? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com
[asterisk-users] include sip configuration from another file in sip.conf
Hi, when i include a sip configuration from another file in my sip.conf using #include /etc/asterisk/sip-sipgate.conf everything seems to be working. The peer is listed when i execute sip show peers and Status is OK. But the peer is not listed using sip show registry. I need to place the register = ... in the sip.conf to make it work. Is this working as expected or is it a bug? Regards Robert Wagner signature.asc Description: OpenPGP digital signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] pattern containing an asterisk
Hi, i need to match a number with like 03012345678*0 or 03012345*9 I tried _X.*X and _X!*X but both are maching 03012345678 too, ignoring that *X is required at the end. The interesting part is that like expected _X*X is matching only numbers like 1*1 and not 11 Regards Robert Wagner signature.asc Description: OpenPGP digital signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunk between two Asterisk servers
What are your allowguest= and domain= settings in the global section of sip.conf? And which version of Asterisk exactly are you using? Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] include sip configuration from another file in sip.conf
On 05/12/2010 01:03 PM, Robert Wagner wrote: Hi, when i include a sip configuration from another file in my sip.conf using #include /etc/asterisk/sip-sipgate.conf everything seems to be working. The peer is listed when i execute sip show peers and Status is OK. But the peer is not listed using sip show registry. I need to place the register = ... in the sip.conf to make it work. Is this working as expected or is it a bug? Working as expected. When you #include a file, the #include line is replaced with the contents of the file. Meaning your register line is likely being placed inside the previous context. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] include sip configuration from another file in sip.conf
On 05/12/2010 01:03 PM, Robert Wagner wrote: when i include a sip configuration from another file in my sip.conf using #include /etc/asterisk/sip-sipgate.conf everything seems to be working. The peer is listed when i execute sip show peers and Status is OK. But the peer is not listed using sip show registry. I need to place the register = ... in the sip.conf to make it work. Is this working as expected or is it a bug? On Wed, 12 May 2010, Jason Parker wrote: Working as expected. When you #include a file, the #include line is replaced with the contents of the file. Meaning your register line is likely being placed inside the previous context. An include file like the following will work as the OP expected: [general](+) register= x:yyy...@sipgate.com/zz [sipgate.com] caninvite = no canreinvite = no context = from-sipgate.com fromdomain = sipgate.com fromuser= x host= sipgate.com insecure= very nat = no secret = yy type= peer username= x -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] bad magic number log messages
Many are having this problem. goto http://issues.asterisk.org and search for 'bad magic number' Notably, a few reports have come up in recent days. Alec Davis _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Rose Sent: Thursday, 13 May 2010 3:00 a.m. To: asterisk-users@lists.digium.com Subject: [asterisk-users] bad magic number log messages Anyone else get this issue - around 200 entries per second of this in the Asterisk messages file: astobj2.c:115 INTERNAL_OBJ: bad magic number 0x27b4113a Seems to happen after several hours of receiving a steady stream of test calls. My messages file is 7.5 gigs. John -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sangoma A101D PRI failing with ERROR - -- Got SABME from network peer. Sending Unnumbered Acknowledgement
Hi Guys, Anyone might know why this error keeps showing up and inbound/outbound is not working on a Bell PRI with Sangoma A101D? -- Got SABME from network peer. Sending Unnumbered Acknowledgement No calls can be made inbound/outbound. Keeps repeating. No alarms ON and no changes been made to the system. Stopped all a sudden. Asterisk CLI doesn't show anything with full verbose for both inbound and outbound. pbx*CLI pri show span 1 Primary D-channel: 24 Status: Provisioned, Up, Active Switchtype: National ISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 1 Retrans: 0 Busy: 0 Overlap Dial: 0 Logical Channel Mapping: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T309 Timer: -1 T313 Timer: 4000 N200 Counter: 3 Thanks, Bruce -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma A101D PRI failing with ERROR - -- Got SABME from network peer. Sending Unnumbered Acknowledgement
- bruce bruce bruceb...@gmail.com wrote: Hi Guys, Anyone might know why this error keeps showing up and inbound/outbound is not working on a Bell PRI with Sangoma A101D? -- Got SABME from network peer. Sending Unnumbered Acknowledgement No calls can be made inbound/outbound. Keeps repeating. No alarms ON and no changes been made to the system. Stopped all a sudden. Asterisk CLI doesn't show anything with full verbose for both inbound and outbound. pbx*CLI pri show span 1 Primary D-channel: 24 Status: Provisioned, Up, Active Switchtype: National ISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 1 Retrans: 0 Busy: 0 Overlap Dial: 0 Logical Channel Mapping: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T309 Timer: -1 T313 Timer: 4000 N200 Counter: 3 --- If it 'just started happening' something must have changed. Have you contacted your telco to confirm there are no line issues for your circuit? Assuming that checks out, are you certain nothing has changed on your Asterisk box? The next step I'd take is to contact Sangoma tech support. Their team is top notch. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 - providers discontinuing support
What is wrong with IAX2 protocol? If IAX2 is so much better than SIP so why providers discontinuing support for IAX2 I was with provider callwithus but they discontinue IAX2 I switched to checkbox.cc but they discontinued it as well. What is wrong with IAX2? -- Joseph -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ringback
Hi, I'm going abroad shortly and want to be able to dial into asterisk and get it to call me back so that I can make an outgoing call through my voip provider, rather than paying crazy international rates. Can anyone point me in the right direction with regards to the dialplan? Im using Asterisk 1.4 Many thanks Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] bad magic number log messages
OK thanks. Yes I see this is reported in 1.6.0.27 which is where I started seeing it. John From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alec Davis Sent: Wednesday, May 12, 2010 1:52 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] bad magic number log messages Many are having this problem. goto http://issues.asterisk.org and search for 'bad magic number' Notably, a few reports have come up in recent days. Alec Davis From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Rose Sent: Thursday, 13 May 2010 3:00 a.m. To: asterisk-users@lists.digium.com Subject: [asterisk-users] bad magic number log messages Anyone else get this issue - around 200 entries per second of this in the Asterisk messages file: astobj2.c:115 INTERNAL_OBJ: bad magic number 0x27b4113a Seems to happen after several hours of receiving a steady stream of test calls. My messages file is 7.5 gigs... John -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] bad magic number log messages
I should have added, that if you havn't already, please report your senario with example dialplan etc to one of the open bug reports related to you problem, otherwise feel free to open a new one. Also 'many' was a bit strong, should have said 'others'. Alec Davis _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alec Davis Sent: Thursday, 13 May 2010 7:52 a.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] bad magic number log messages Many are having this problem. goto http://issues.asterisk.org and search for 'bad magic number' Notably, a few reports have come up in recent days. Alec Davis _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Rose Sent: Thursday, 13 May 2010 3:00 a.m. To: asterisk-users@lists.digium.com Subject: [asterisk-users] bad magic number log messages Anyone else get this issue - around 200 entries per second of this in the Asterisk messages file: astobj2.c:115 INTERNAL_OBJ: bad magic number 0x27b4113a Seems to happen after several hours of receiving a steady stream of test calls. My messages file is 7.5 gigs. John -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problems with unicall
Hello, i'm using asterisk 1.4.9 in fedora 7, i was compiled its with this package: libpri-1.4.2 asterisk-1.4.9 spandsp-0.0.4 unicall-0.0.5pre1 libmfcr2-0.0.3 libsupertone-0.0.2 libunicall-0.0.3 zaptel-1.4.4 i'm using a E1 pci card with R2 but they not work, when I start the asterisk its generate this log: [May 12 08:53:24] WARNING[30814] channel.c: No channel type registered for 'Unicall' [May 12 08:53:24] WARNING[30814] app_dial.c: Unable to create channel of type 'Unicall' (cause 66 - Channel not implemented) [May 12 08:54:47] NOTICE[2613] cdr.c: CDR simple logging enabled. [May 12 08:54:47] NOTICE[2613] loader.c: 146 modules will be loaded. [May 12 08:54:49] WARNING[2613] res_smdi.c: No SMDI interfaces are available to listen on, not starting SDMI listener. [May 12 08:54:50] WARNING[2613] chan_sip.c: insecure=very at line 37 is deprecated; use insecure=port,invite instead [May 12 08:54:50] WARNING[2613] chan_zap.c: Unable to specify channel 1: Device or resource busy [May 12 08:54:50] ERROR[2613] chan_zap.c: Unable to open channel 1: Device or resource busy here = 0, tmp-channel = 1, channel = 1 [May 12 08:54:50] ERROR[2613] chan_zap.c: Unable to register channel '1-15,17-31' [May 12 08:54:50] NOTICE[2613] pbx_ael.c: Starting AEL load process. [May 12 08:54:50] NOTICE[2613] pbx_ael.c: AEL load process: calculated config file name '/etc/asterisk/extensions.ael'. [May 12 08:54:50] WARNING[2613] ael.y: File: /etc/asterisk/extensions.ael, Line 112, Cols: 34-34: Warning! The empty context ael-dundi-e164-canonical will be IGNORED! [May 12 08:54:50] WARNING[2613] ael.y: File: /etc/asterisk/extensions.ael, Line 120, Cols: 34-34: Warning! The empty context ael-dundi-e164-customers will be IGNORED! [May 12 08:54:50] WARNING[2613] ael.y: File: /etc/asterisk/extensions.ael, Line 128, Cols: 33-33: Warning! The empty context ael-dundi-e164-via-pstn will be IGNORED! [May 12 08:54:50] NOTICE[2613] pbx_ael.c: AEL load process: parsed config file name '/etc/asterisk/extensions.ael' . [May 12 08:54:50] WARNING[2613] pbx_ael.c: Warning: file /etc/asterisk/extensions.ael, line 141-145: The included context 'ael-dundi-e164-canonical' cannot be found. [May 12 08:54:50] WARNING[2613] pbx_ael.c: Warning: file /etc/asterisk/extensions.ael, line 141-145: The included context 'ael-dundi-e164-customers' cannot be found. [May 12 08:54:50] WARNING[2613] pbx_ael.c: Warning: file /etc/asterisk/extensions.ael, line 141-145: The included context 'ael-dundi-e164-via-pstn' cannot be found. [May 12 08:54:50] WARNING[2613] pbx_ael.c: Warning: file /etc/asterisk/extensions.ael, line 276-283: The included context 'ael-parkedcalls' cannot be found. [May 12 08:54:50] NOTICE[2613] pbx_ael.c: AEL load process: checked config file name '/etc/asterisk/extensions.ael'. [May 12 08:54:50] NOTICE[2613] pbx_ael.c: AEL load process: compiled config file name '/etc/asterisk/extensions.ael'. [May 12 08:54:50] NOTICE[2613] pbx_ael.c: AEL load process: merged config file name '/etc/asterisk/extensions.ael'. [May 12 08:54:50] WARNING[2613] pbx.c: Context 'ael-local' tries includes nonexistent context 'ael-parkedcalls' my file unicall.conf is this: [channels] context=e1-inline usecallerid=yes hidecallerid=no threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 relaxdtmf=yes callgroup=1 pickupgroup=1 immediate=no callerid=asreceived musiconhold=default protocolclass=mfcr2 protocolvariant=br,20,20,20 protocolend=cpe group=1 channel = 1-15 channel = 17-31 and when i do any dial by asterisk return this: The 'dial' command is deprecated and will be removed in a future release. Please use 'console dial' instead. == Console is full duplex -- Executing [...@ext-local:1] Dial(OSS/dsp, Unicall/g1/32719595) in new stack [May 12 08:58:43] WARNING[2689]: chan_unicall.c:1034 unicall_call: Make call failed - Blocked -- Couldn't call g1/32719595 -- Hungup 'UniCall/1-1' == Everyone is busy/congested at this time (0:0/0/0) == Auto fallthrough, channel 'OSS/dsp' status is 'CHANUNAVAIL' can somebody help me? Sorry for my english. Marcelo Nunes Dos santos -- TI Savarsul - nu...@savarsul.com.br Blog: makelinux.com.br Email/MSN: marcelo7...@gmail.com twitter: www.twitter.com/marcelonunes -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 - providers discontinuing support
What is wrong with IAX2 protocol? If IAX2 is so much better than SIP so why providers discontinuing support for IAX2 I was with provider callwithus but they discontinue IAX2 I switched to checkbox.cc but they discontinued it as well. What is wrong with IAX2? The same thing that's wrong with a lot of theoretically superior technologies: SIP is *more* universal, and therefore if it's a choice of supporting two technologies or just one, SIP has more bang for the buck. Almost every gadget or gizmo supports SIP. Few support IAX2. To support IAX2 for the relatively small number of people who know what it is and who are running Asterisk or IAX2-capable gear may be more trouble than it is worth. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunk between two Asterisk servers
--- On Wed, 5/12/10, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: What are your allowguest= and domain= settings in the global section of sip.conf? And which version of Asterisk exactly are you using? I have no such settings defined yet. Still haven't tried to set them... Not sure what to put in domain. Anyway: # /etc/asterisk/sip.conf [general] vmexten=*97 disallow=all allow=ulaw allow=alaw context=from-sip-external callerid=Unknown notifyringing=yes notifyhold=yes limitonpeers=yes tos_sip=cs3 tos_audio=ef tos_video=af41 rtptimeout=120 rtpholdtimeout=300 pedantic=no urlencode=yes register=01:...@internet_sip_provider.com/01010101010101 regcontext=dundi-extens Server 2: Asterisk 1.4.31 Server 1: same sip.conf settings except Asterisk 1.2.40 Notice the urlencode setting which is a patch taken from: https://issues.asterisk.org/view.php?id=14652 This may be the culprit but I'm not quite sure about it. Also, I *need* this patch unless the address incomplete issue gets solved. Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunk between two Asterisk servers
--- On Wed, 5/12/10, Vardan hvarda...@gmail.com wrote: Please change the peers name in any server. for example: server1: interboxsip1 server2: interboxsip2 If I understand correctly, the peer names can be identical on both servers. What counts is the host entry, I guess. But then again, my SIP trunk isn't working so I'll try out your suggestion tomorrow. Thanks, Vieri Vardan Vieri wrote: --- On Wed, 5/12/10, Vardanhvarda...@gmail.com wrote: please show sip show users and sip show peers SERVER 2: sip show users (trimmed to just my sip test trunk): Username Secret Accountcode Def.Context ACL NAT interboxsip mycontext No RFC3581 sip show peers (also trimmed): Name/username Host Dyn Nat ACL Port Status sipprovider/01 w.x.y.z N 5060 OK (90 ms) interboxsip 192.168.250.111 5060 Unmonitored 7503/7503 10.215.146.190 D N A 5060 OK (20 ms) 7502/7502 10.215.146.203 D N A 5060 OK (20 ms) 7172/7172 192.168.250.7 D N A 13404 OK (40 ms) 7166/7166 10.215.146.200 D N A 5060 OK (20 ms) 7165/7165 10.215.248.12 D N A 5060 OK (1 ms) 7160/7160 10.215.146.182 D N A 5060 OK (20 ms) 7137/7137 192.168.250.6 D N A 25967 OK (10 ms) 7118/7118 192.168.250.10 D N A 14508 OK (1 ms) 7117/7117 10.215.146.185 D N A 5060 OK (20 ms) 7114/7114 192.168.250.8 D N A 12342 OK (10 ms) 7112/7112 192.168.250.31 D N A 19829 OK (10 ms) 7111/7111 192.168.250.32 D N A 35259 OK (80 ms) 7109/7109 (Unspecified) D N A 0 UNKNOWN 7097/7097 10.215.146.164 D N A 5060 OK (20 ms) SERVER 1: sip show users is identical. sip show peers (trimmed): Name/username Host Dyn Nat ACL Port Status sipprovider/01 w.x.y.z N 5060 OK (79 ms) interboxsip 192.168.250.112 5060 Unmonitored vardan Vieri wrote: --- On Wed, 5/12/10, Philipp von Klitzingklitz...@pool.informatik.rwth-aachen.de wrote: --- SIP read from 192.168.250.111:5060 --- SIP/2.0 407 Proxy Authentication Required You need to run the SIP debug on 192.168.250.111 to learn more about WHY the 407 is issued. Have a close look and you are likely to understand it right away. Also: Do not forget the reload after applying changes to sip.conf. I always do a sip reload after changes to sip settings. Here are the SIP messages on 192.168.250.111 (Asterisk server 1 - receiving end): -- SIP read from 192.168.250.112:5060: INVITE sip:3...@192.168.250.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport From: devicesip:4...@192.168.250.112;tag=as18a568d6 To:sip:3...@192.168.250.111 Contact:sip:4...@192.168.250.112 Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 12 May 2010 09:20:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO upported: replaces Content-Type: application/sdp Content-Length: 270 v=0 o=root 20611 20611 IN IP4 192.168.250.112 s=session c=IN IP4 192.168.250.112 t=0 0 m=audio 14648 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- (14 headers 13 lines) --- Using INVITE request as basis request - 328617546726e5d430538e8061771...@192.168.250.112 Sending to 192.168.250.112 : 5060 (NAT) Reliably Transmitting (NAT) to 192.168.250.112:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;received=192.168.250.112;rport=5060 From: devicesip:4...@192.168.250.112;tag=as18a568d6 To:sip:3...@192.168.250.111;tag=as57a19dac Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=1327c5b6 Content-Length: 0 --- Scheduling destruction of call '328617546726e5d430538e8061771...@192.168.250.112' in 15000 ms Found user '4053' -- SIP read from 192.168.250.112:5060: ACK sip:3...@192.168.250.111 SIP/2.0 Via: SIP/2.0/UDP
Re: [asterisk-users] IAX2 - providers discontinuing support
SIP is just more supported so its easier for the providers to deal with it. I personally also believe that IAX is not supported by big providers because if they do so, it'll just make asterisk more famous than they want it to be. Secondly, as IAX name suggests, it was primarily designed for trunking between asterisk servers, and it does it with less headaches than SIP, given that your provider supports it well. Zeeshan A Zakaria -- Sent from my Android phone with K-9 Mail. On 2010-05-12 5:12 PM, Joe Greco jgr...@ns.sol.net wrote: What is wrong with IAX2 protocol? If IAX2 is so much better than SIP so why providers discontinu... The same thing that's wrong with a lot of theoretically superior technologies: SIP is *more* universal, and therefore if it's a choice of supporting two technologies or just one, SIP has more bang for the buck. Almost every gadget or gizmo supports SIP. Few support IAX2. To support IAX2 for the relatively small number of people who know what it is and who are running Asterisk or IAX2-capable gear may be more trouble than it is worth. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. -- _ -- Bandwidth and Colocati... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 - providers discontinuing support
On 05/12/10 16:04, Joe Greco wrote: What is wrong with IAX2 protocol? If IAX2 is so much better than SIP so why providers discontinuing support for IAX2 I was with provider callwithus but they discontinue IAX2 I switched to checkbox.cc but they discontinued it as well. What is wrong with IAX2? The same thing that's wrong with a lot of theoretically superior technologies: SIP is *more* universal, and therefore if it's a choice of supporting two technologies or just one, SIP has more bang for the buck. Almost every gadget or gizmo supports SIP. Few support IAX2. To support IAX2 for the relatively small number of people who know what it is and who are running Asterisk or IAX2-capable gear may be more trouble than it is worth. ... JG Any recommendation for reliable IAX2 provider with good rates to Asia? -- Joseph -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pattern containing an asterisk
At 8:04 PM on 12 May 2010, Robert Wagner wrote: i need to match a number with like 03012345678*0 or 03012345*9 I tried _X.*X and _X!*X but both are maching 03012345678 too, ignoring that *X is required at the end. The interesting part is that like expected _X*X is matching only numbers like 1*1 and not 11 The . in a pattern is meant only to be used at the end, to match any remaining characters. The *X after the dot in your pattern is just being ignored. Have you checked for warnings in your log? I'm not sure if Asterisk issues a warning on that, but I think it should. One thing you could do is make one pattern for each possible length. e.g.: _XXX*X and _*X If you need it to be variable length, I think you would need to use the Read application instead of standard dialplan matching. -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 signature.asc Description: PGP signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simulating a commercial SIP provider
Quoting Alfredo Peña arp...@gmail.com: Try using this line in the [general] section of sip.conf in your simulated SIP provider machine: realm=sip.provider.com No, that didn't seem to make any difference. However, this did: insecure=invite This prevents the Failed to authenticate on INVITE errors from occurring on both sides when INVITE messages arrive with user names (before the @ sign) that are only known on the remote system. The user names are associated with the phones that I use on either end of the connection. Unfortunately, I'm forced to use this option on both sides of the connection, instead of only on the provider side. Therefore, it's still not really the answer that I'm looking for, but it's a step in the right direction. Cheers, Jaap -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk core dumping on SendFax with FFA
Well, I have managed to get my hands on a copy of 1.2.1 rc1 FFA which seems to have fixed the core dumping issue but does not appear to have fixed the issue that was causing the core dump. We are still getting an issue with a particular file which I have tried multiple different ways to create to no avail. The tiff file is created with ghostscript from a pdf as per the guidlines but every time we try and fax it we get the following: [May 13 11:28:09] DEBUG[26959] res_fax_digium.c: FAX handle 0: created document queue [May 13 11:28:09] ERROR[26959] res_fax_digium.c: FAX handle 0: failed to queue document '/var/spool/asterisk/fax/campaign_70.tif' [May 13 11:28:09] DEBUG[26959] res_fax_digium.c: FAX handle 0: freeing document queue. [May 13 11:28:09] DEBUG[26959] res_fax_digium.c: FAX handle 0: closing [May 13 11:28:09] ERROR[26959] res_fax.c: channel 'SIP/teleblast-sbc01-0001' FAX session '1' failure, reason: 'failed to start F AX session' and the call terminates. tiffinfo for the file shows: TIFF Directory at offset 0x2aae0 (174816) Subfile Type: multi-page document (2 = 0x2) Image Width: 1767 Image Length: 2369 Resolution: 204, 196 pixels/inch Bits/Sample: 1 Compression Scheme: CCITT Group 4 Photometric Interpretation: min-is-white FillOrder: msb-to-lsb Orientation: row 0 top, col 0 lhs Samples/Pixel: 1 Rows/Strip: 2369 Planar Configuration: single image plane Page Number: 0-0 Software: GPL Ghostscript 8.70 DateTime: 2010:05:12 23:20:00 Group 4 Options: (0 = 0x0) TIFF Directory at offset 0x57172 (356722) Subfile Type: multi-page document (2 = 0x2) Image Width: 1767 Image Length: 2369 Resolution: 204, 196 pixels/inch Bits/Sample: 1 Compression Scheme: CCITT Group 4 Photometric Interpretation: min-is-white FillOrder: msb-to-lsb Orientation: row 0 top, col 0 lhs Samples/Pixel: 1 Rows/Strip: 2369 Planar Configuration: single image plane Page Number: 1-0 Software: GPL Ghostscript 8.70 DateTime: 2010:05:12 23:20:11 Group 4 Options: (0 = 0x0) And the file is 357026 bytes in size. Can anyone see anything wrong with the tiff info or does anyone know of any issues with multiple pages or file size with fax for asterisk? Unfortunately I cannot seem to find any more information as to why the document couldn't be queued. Cheers, Ben -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk core dumping on SendFax with FFA
On Wed, May 12, 2010 at 9:53 PM, Ben Dinnerville b...@voicelogic.com.au wrote: We are still getting an issue with a particular file which I have tried multiple different ways to create to no avail. The tiff file is created with ghostscript from a pdf as per the guidlines but every time we try and fax it we get the following: Try using ReceiveFax(), and use a pdf=to-fax service or your own fax machine to fax it to your asterisk machine. That will give you an asterisk-created tiff of your pdf. I used ReceiveFax() to generate the tiffs I use for SendFax() testing. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problems with unicall
I already replied to you in the asterisk-r2 mailing list. Your lines are blocked, the log is telling you that: [May 12 08:58:43] WARNING[2689]: chan_unicall.c:1034 unicall_call: Make call failed - Blocked The only way you get that is if the line is blocked ( rx ABCD bits are 1101 or equivalent blocked for your country ) or the line is configured only for incoming calls ( not possible since chan_unicall.c hard-codes that parameter to allow calls in both ways ). Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com On Wed, May 12, 2010 at 5:03 PM, Marcelo nunes dos santos marcelo7...@gmail.com wrote: Hello, i'm using asterisk 1.4.9 in fedora 7, i was compiled its with this package: libpri-1.4.2 asterisk-1.4.9 spandsp-0.0.4 unicall-0.0.5pre1 libmfcr2-0.0.3 libsupertone-0.0.2 libunicall-0.0.3 zaptel-1.4.4 i'm using a E1 pci card with R2 but they not work, when I start the asterisk its generate this log: [May 12 08:53:24] WARNING[30814] channel.c: No channel type registered for 'Unicall' [May 12 08:53:24] WARNING[30814] app_dial.c: Unable to create channel of type 'Unicall' (cause 66 - Channel not implemented) [May 12 08:54:47] NOTICE[2613] cdr.c: CDR simple logging enabled. [May 12 08:54:47] NOTICE[2613] loader.c: 146 modules will be loaded. [May 12 08:54:49] WARNING[2613] res_smdi.c: No SMDI interfaces are available to listen on, not starting SDMI listener. [May 12 08:54:50] WARNING[2613] chan_sip.c: insecure=very at line 37 is deprecated; use insecure=port,invite instead [May 12 08:54:50] WARNING[2613] chan_zap.c: Unable to specify channel 1: Device or resource busy [May 12 08:54:50] ERROR[2613] chan_zap.c: Unable to open channel 1: Device or resource busy here = 0, tmp-channel = 1, channel = 1 [May 12 08:54:50] ERROR[2613] chan_zap.c: Unable to register channel '1-15,17-31' [May 12 08:54:50] NOTICE[2613] pbx_ael.c: Starting AEL load process. [May 12 08:54:50] NOTICE[2613] pbx_ael.c: AEL load process: calculated config file name '/etc/asterisk/extensions.ael'. [May 12 08:54:50] WARNING[2613] ael.y: File: /etc/asterisk/extensions.ael, Line 112, Cols: 34-34: Warning! The empty context ael-dundi-e164-canonical will be IGNORED! [May 12 08:54:50] WARNING[2613] ael.y: File: /etc/asterisk/extensions.ael, Line 120, Cols: 34-34: Warning! The empty context ael-dundi-e164-customers will be IGNORED! [May 12 08:54:50] WARNING[2613] ael.y: File: /etc/asterisk/extensions.ael, Line 128, Cols: 33-33: Warning! The empty context ael-dundi-e164-via-pstn will be IGNORED! [May 12 08:54:50] NOTICE[2613] pbx_ael.c: AEL load process: parsed config file name '/etc/asterisk/extensions.ael' . [May 12 08:54:50] WARNING[2613] pbx_ael.c: Warning: file /etc/asterisk/extensions.ael, line 141-145: The included context 'ael-dundi-e164-canonical' cannot be found. [May 12 08:54:50] WARNING[2613] pbx_ael.c: Warning: file /etc/asterisk/extensions.ael, line 141-145: The included context 'ael-dundi-e164-customers' cannot be found. [May 12 08:54:50] WARNING[2613] pbx_ael.c: Warning: file /etc/asterisk/extensions.ael, line 141-145: The included context 'ael-dundi-e164-via-pstn' cannot be found. [May 12 08:54:50] WARNING[2613] pbx_ael.c: Warning: file /etc/asterisk/extensions.ael, line 276-283: The included context 'ael-parkedcalls' cannot be found. [May 12 08:54:50] NOTICE[2613] pbx_ael.c: AEL load process: checked config file name '/etc/asterisk/extensions.ael'. [May 12 08:54:50] NOTICE[2613] pbx_ael.c: AEL load process: compiled config file name '/etc/asterisk/extensions.ael'. [May 12 08:54:50] NOTICE[2613] pbx_ael.c: AEL load process: merged config file name '/etc/asterisk/extensions.ael'. [May 12 08:54:50] WARNING[2613] pbx.c: Context 'ael-local' tries includes nonexistent context 'ael-parkedcalls' my file unicall.conf is this: [channels] context=e1-inline usecallerid=yes hidecallerid=no threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 relaxdtmf=yes callgroup=1 pickupgroup=1 immediate=no callerid=asreceived musiconhold=default protocolclass=mfcr2 protocolvariant=br,20,20,20 protocolend=cpe group=1 channel = 1-15 channel = 17-31 and when i do any dial by asterisk return this: The 'dial' command is deprecated and will be removed in a future release. Please use 'console dial' instead. == Console is full duplex -- Executing [...@ext-local:1] Dial(OSS/dsp, Unicall/g1/32719595) in new stack [May 12 08:58:43] WARNING[2689]: chan_unicall.c:1034 unicall_call: Make call failed - Blocked -- Couldn't call g1/32719595 -- Hungup 'UniCall/1-1' == Everyone is busy/congested at this time (0:0/0/0) == Auto fallthrough, channel 'OSS/dsp' status is 'CHANUNAVAIL' can somebody help me? Sorry for my english.
[asterisk-users] Error at start of asterisk with cdr_addon_mysql.o
Hi all,
I use asterisk-1.6.2.7 and asterisk addon version 1.6.2.1.
It started ok with out cdr_addon_mysql.o. But when I put
cdr_addon_mysql.o in to modules folder, it fail at start and the
following out has been thrown:
--
[r...@localhost modules]# /usr/sbin/safe_asterisk: line 145: 13270
Segmentation fault (core dumped) nice -n $PRIORITY ${ASTBINDIR}/asterisk
-f ${CLIARGS} ${ASTARGS} /dev/${TTY} 21 /dev/${TTY}
Asterisk exited with exit status 139
Asterisk exited on signal 11
Automatically restarting Asterisk.
---
What is the problem?
Thanks in advance.
Quyps
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Re: [asterisk-users] Error at start of asterisk with cdr_addon_mysql.o
On Thu, 13 May 2010, Pham Quy wrote:
Hi all,
I use asterisk-1.6.2.7 and asterisk addon version 1.6.2.1.
It started ok with out cdr_addon_mysql.o. But when I put
cdr_addon_mysql.o in to modules folder, it fail at start and the
following out has been thrown:
--
[r...@localhost modules]# /usr/sbin/safe_asterisk: line 145: 13270
Segmentation fault (core dumped) nice -n $PRIORITY ${ASTBINDIR}/asterisk
-f ${CLIARGS} ${ASTARGS} /dev/${TTY} 21 /dev/${TTY}
Asterisk exited with exit status 139
Asterisk exited on signal 11
Automatically restarting Asterisk.
---
What is the problem?
The problem is...
You have no clue[s] :)
First off, the module should be cdr_addon_mysql.so, not cdr_addon_mysql.o.
If you don't have the so in /usr/lib/asterisk/modules/ something is
wrong with your build.
Try something like this:
sudo -u whatever-user-runs-asterisk-on-your-system\
/usr/sbin/asterisk -c -d -d -d -f -g -n -v -v -v
Or, you can start Asterisk without loading cdr_addon_mysql.so and then
load it from the Asterisk CLI. It sounds like you are auto-loading modules
so you could add noload=cdr_addon_mysql.so to
/etc/asterisk/modules.conf to get Asterisk running and then load it with
something like load cdr_addon_mysql.so
I'm a 1.2 Luddite so the commands may have changed slightly. Also,
depending on the specifics of your installation, the paths may be
different.
See if this gives you any clues.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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