Re: [asterisk-users] T38 negotiation, the last step !
Sorry for responding so late. Does this mean that the fax is going to a dialpeer that isn't being configured to handle T.38 ? I'm trying to send the faxes with just 1 entire number, so I don0t think the call is not matching that dialpeer . . . . any thoughts ? On Fri, Jul 17, 2009 at 3:33 PM, Steve Underwood wrote: > Klaus Darilion wrote: > > Xavier Cardil schrieb: > > > >> Hi, I've managed to get HYLAFAX>T38MODEM-> > >> ASTERISK>CISCOAS5400 working, but when they are negotiating asterisk > >> drops a message telling "Unknown RTP codec 96 received from gateway" Do > >> somebody know how to fix it ? > >> > >> Thank you ! > >> > >> > >> > >> << [ TYPE: Control (4) SUBCLASS: Ringing (3) ] > [SIP/GWCISCO5400O-600bfcc8] > >> << [ TYPE: Control (4) SUBCLASS: Answer (4) ] > [SIP/GWCISCO5400O-600bfcc8] > >> << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] > [SIP/GWCISCO5400O-600bfcc8] > >> << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] > [SIP/GWCISCO5400O-600bfcc8] > >> << [ TYPE: Control (4) SUBCLASS: T38/Negotiation Requested (19) ] > >> [SIP/GWCISCO5400O-600bfcc8] > >> << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] > [SIP/GWCISCO5400O-600bfcc8] > >> [Jul 16 17:50:39] NOTICE[2736]: rtp.c:1739 ast_rtp_read: Unknown RTP > >> codec 96 received from '192.168.3.163' > >> > > > > Is '192.168.3.163' the Cisco GW? If yes, then it is probably an NSE > > packet. NSE is a Cisco proprietary FaxoverIP solution and uses per > > default payload types 96 and 97 to signal a changeover from VoIP to FoIP. > > > > Probably you have to configure the Cisco GW to use T.38 instead of NSE > > for FoIP. > > > When did RFC2833 become a proprietary Cisco spec? > > The NSEs just signal the startup of FAX. You still need to switch into > T.38 to actually exchange the FAX. > > Steve > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T38 negotiation, the last step !
Hi Kelvin, thank you for your response, well in fact it is not working but that's only a NOTICE, not an error. Warnings comes after that and the fax is not sent. Take a look at the last lines of this output : << [ TYPE: Control (4) SUBCLASS: Ringing (3) ] [SIP/GWCISCO5400O-01ba5638] IVR3*CLI> debug channel allSIP/GWCISCO5400O-01ba5638 SIP/T38modem-01ba2178 << [ TYPE: Control (4) SUBCLASS: Answer (4) ] [SIP/GWCISCO5400O-01ba5638] << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-01ba5638] << [ TYPE: Control (4) SUBCLASS: T38/Negotiation Requested (19) ] [SIP/GWCISCO5400O-01ba5638] << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-01ba5638] [Jul 16 18:20:20] NOTICE[2846]: rtp.c:1739 ast_rtp_read: Unknown RTP codec 96 received from '192.168.3.163' << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-01ba5638] [Jul 16 18:20:21] NOTICE[2846]: rtp.c:1739 ast_rtp_read: Unknown RTP codec 96 received from '192.168.3.163' << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-01ba5638] [Jul 16 18:20:21] NOTICE[2846]: rtp.c:1739 ast_rtp_read: Unknown RTP codec 96 received from '192.168.3.163' << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-01ba5638] << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-01ba5638] << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-01ba5638] [Jul 16 18:20:30] WARNING[2815]: chan_sip.c:3075 retrans_pkt: Maximum retries exceeded on transmission 1af1c92e-9270-de11-8264-001517bb9...@ivr3for seqno 1 (Critical Response) -- See doc/sip-retransmit.txt. [Jul 16 18:20:30] WARNING[2815]: chan_sip.c:3102 retrans_pkt: Hanging up call 1af1c92e-9270-de11-8264-001517bb9...@ivr3 - no reply to our critical packet (see doc/sip-retransmit.txt). Do you have an idea of what is happening ? I sniffed UDP traffic on port 5060 and I get : INVITE SDP ( t38 ) --> TRYING <-- NOT ACCEPTABLE HERE< ACK> BYE<--- 200 OK--------> Thanks for your help. On Thu, Jul 16, 2009 at 6:17 PM, Kevin P. Fleming wrote: > Xavier Cardil wrote: > > Hi, I've managed to get HYLAFAX>T38MODEM-> > > ASTERISK>CISCOAS5400 working, but when they are negotiating asterisk > > drops a message telling "Unknown RTP codec 96 received from gateway" Do > > somebody know how to fix it ? > > There's nothing to fix; the gateway sent an expected RTP packet, which > Asterisk dropped. If your FAX works, then you can ignore this. > > -- > Kevin P. Fleming > Digium, Inc. | Director of Software Technologies > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > skype: kpfleming | jabber: kpflem...@digium.com > Check us out at www.digium.com & www.asterisk.org > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > graph1 Description: Binary data ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] T38 negotiation, the last step !
Hi, I've managed to get HYLAFAX>T38MODEM->ASTERISK>CISCOAS5400 working, but when they are negotiating asterisk drops a message telling "Unknown RTP codec 96 received from gateway" Do somebody know how to fix it ? Thank you ! << [ TYPE: Control (4) SUBCLASS: Ringing (3) ] [SIP/GWCISCO5400O-600bfcc8] << [ TYPE: Control (4) SUBCLASS: Answer (4) ] [SIP/GWCISCO5400O-600bfcc8] << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8] << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8] << [ TYPE: Control (4) SUBCLASS: T38/Negotiation Requested (19) ] [SIP/GWCISCO5400O-600bfcc8] << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8] [Jul 16 17:50:39] NOTICE[2736]: rtp.c:1739 ast_rtp_read: Unknown RTP codec 96 received from '192.168.3.163' << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8] [Jul 16 17:50:40] NOTICE[2736]: rtp.c:1739 ast_rtp_read: Unknown RTP codec 96 received from '192.168.3.163' << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8] [Jul 16 17:50:41] NOTICE[2736]: rtp.c:1739 ast_rtp_read: Unknown RTP codec 96 received from '192.168.3.163' << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8] << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8] << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk + cisco as5400 t.38 fax sending.
Hello, I heard that since asterisk 1.6.0.6, now you can send faxes with t.38 through asterisk to a PST gateway that supports t.38 too. Is that true ? If so, what elements you need to make it work beside asterisk and the PSTN trunk ? Thanks all.- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk capacity
You can handle 600 SIP sessions and about 400 calls doing transcoding ( passing RTP ) On Mon, Jul 6, 2009 at 1:26 PM, Steve Totaro wrote: > It can make 9977.39 Bogocalls of course! > > On Mon, Jul 6, 2009 at 5:17 AM, abdelkader > wrote: > > Hello, > > > > This is the configuration of my server got from PHP system info: > > > > System Vital: > > Kernel Version 2.6.18-6-amd64 (SMP) > > Distro Name Debian 4.0 > > > > Hardware Information: > > Processors 4 > > Model Intel(R) Xeon(R) CPU E5420 @ 2.50GHz > > CPU Speed 2.49 GHz > > Cache Size 6.00 MB > > System Bogomips 19954.78 > > PCI Devices none > > IDE Devices none > > SCSI Devices > > -DELL PERC 6/i (Direct-Access) > > -DP BACKPLANE (Enclosure) > > -TSSTcorp DVD-ROM TS-L333A (CD-ROM) > > USB Devices > > -Dell Computer Corp. > > -Cypress Semiconductor Corp. CY7C65640 USB-2.0 "TetraHub" > > -Dell Computer Corp. Hub > > > > Mounted Filesystems: > > Mount Type Partition Percent Capacity Free Used Size > > / ext3 /dev/sda1 0% 759.28 GB 1.63 GB 801.63 GB > > /dev/shm tmpfs tmpfs 0% (1%) 3.90 GB 0.00 KB 3.90 GB > > /lib/init/rw tmpfs tmpfs 0% (1%) 3.90 GB 0.00 KB 3.90 GB > > /dev tmpfs udev 1% (1%) 9.95 MB 52.00 KB 10.00 MB > > Totals :0% 763.19 GB 1.63 GB 805.54 GB > > > > > > > > > > Memory Usage: > > Type Percent Capacity Free Used Size > > Physical Memory 5% 7.37 GB 437.23 MB 7.80 GB > > - Kernel + applications 2% 194.45 MB > > - Buffers 2% 159.57 MB > > - Cached 1% 83.21 MB > > Disk Swap 0% 22.84 GB 0.00 KB 22.84 GB > > > > > > > > The version of Asterisk is: 1.4.22. > > > > I need to know how many calls I can handle with my Asterisk. > > > > Thks. > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > ___ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > -- > Thanks, > Steve Totaro > +18887771888 (Toll Free) > +12409381212 (Cell) > +12024369784 (Skype) > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: Unknown udp ports listening experts calling !
Hi Bruce, thank you for your recommendations . . . I passed the test and the only wanrning is this one : /usr/sbin/unhide [ Warning ] /usr/sbin/useradd[ OK ] /usr/sbin/userdel[ OK ] /usr/sbin/usermod[ OK ] /usr/sbin/vipw [ OK ] /usr/sbin/unhide-linux26 [ Warning ] On Wed, Jul 1, 2009 at 1:42 PM, Bruce Ferrell wrote: > > > Xavier Cardil wrote: > > I found nothing is passing through those ports . . . I think something > > was sending the stream to our PST/SIP gateways, so the calls where > > affected when getting in to the gateways. I found we are not running any > > extra TCL applications on those gateways . . . could it be possible ? > > Could an UDP stream get mixed with another through an UDP port ? Is a > > very strange issue but I really want to know why . . . any more hints ? > > > > Thanks. > > > > On Wed, Jul 1, 2009 at 11:48 AM, John A. Sullivan III > > mailto:jsulli...@opensourcedevel.com>> > > wrote: > > > > On Wed, 2009-07-01 at 10:14 +0100, Steve Howes wrote: > > > On 1 Jul 2009, at 09:54, Xavier Cardil wrote: > > > > udp0 0 0.0.0.0:2727 <http://0.0.0.0:2727> > > > > 0.0.0.0:* 4989/asterisk > > > > udp0 0 0.0.0.0:9001 <http://0.0.0.0:9001> > > > > 0.0.0.0:* 26354/udp-sender > > > > udp0 0 0.0.0.0:5000 <http://0.0.0.0:5000> > > > > 0.0.0.0:* 4989/asterisk > > > > > > 2727 = mgcp > > > > > > I found that with Google. A useful tool. > > > > I thought 9001 was for JetDirect style print servers. I don't recall > > off the top of my head if they are tcp or udp - John > > -- > > John A. Sullivan III > > Open Source Development Corporation > > +1 207-985-7880 > > jsulli...@opensourcedevel.com <mailto:jsulli...@opensourcedevel.com> > > > > http://www.spiritualoutreach.com > > Making Christianity intelligible to secular society > > > > > Assuming first your box doesn't have a rootkit installed (to check for > a rootkit, use rkhunter. Your distro may have it packaged, if not > google for it) I use lsof to find out what is listening to TCP and UDP > ports: > > lsof -P | grep UDP > lsof -P | grep TCP > > YMMV > > Bruce > > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: Unknown udp ports listening experts calling !
I found nothing is passing through those ports . . . I think something was sending the stream to our PST/SIP gateways, so the calls where affected when getting in to the gateways. I found we are not running any extra TCL applications on those gateways . . . could it be possible ? Could an UDP stream get mixed with another through an UDP port ? Is a very strange issue but I really want to know why . . . any more hints ? Thanks. On Wed, Jul 1, 2009 at 11:48 AM, John A. Sullivan III < jsulli...@opensourcedevel.com> wrote: > On Wed, 2009-07-01 at 10:14 +0100, Steve Howes wrote: > > On 1 Jul 2009, at 09:54, Xavier Cardil wrote: > > > udp0 0 0.0.0.0:2727 > > > 0.0.0.0:* 4989/asterisk > > > udp0 0 0.0.0.0:9001 > > > 0.0.0.0:* 26354/udp-sender > > > udp0 0 0.0.0.0:5000 > > > 0.0.0.0:* 4989/asterisk > > > > 2727 = mgcp > > > > I found that with Google. A useful tool. > > I thought 9001 was for JetDirect style print servers. I don't recall > off the top of my head if they are tcp or udp - John > -- > John A. Sullivan III > Open Source Development Corporation > +1 207-985-7880 > jsulli...@opensourcedevel.com > > http://www.spiritualoutreach.com > Making Christianity intelligible to secular society > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fwd: Unknown udp ports listening experts calling !
-- Forwarded message -- From: Xavier Cardil Date: Wed, Jul 1, 2009 at 10:51 AM Subject: Unknown udp ports listening experts calling ! To: asterisk-users-requ...@lists.digium.com Hello, last days we run under an very heavy issue with one audio stream getting mixed with our RTP traffic. The audio source was unknown and changing the asterisks to other net interfaces and hooking them to another Vlan did the trick. The audio stream is not coming anymore so it is some outside UDP source sending data to that interface. On the way, we changed the asterisk UDP port range to 3-4 instead of the default 1-2. Can somebody tell me why asterisk still listening or transfering data through these ports ? I'm trying to solve the problem, as I find it very interesting. udp0 0 0.0.0.0:27270.0.0.0:* 4989/asterisk udp0 0 0.0.0.0:90010.0.0.0:* 26354/udp-sender udp0 0 0.0.0.0:50000.0.0.0:* 4989/asterisk Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] unwanted locution
No, it doesn't sound like any of our IVR applications. We have 2 two Cisco AS5400 PSTN to SIP gateways, so yes we do outbound / inbound calls. Thank you. On Mon, Jun 29, 2009 at 12:52 PM, Steve Howes wrote: > > On 29 Jun 2009, at 11:00, Xavier Cardil wrote: > > > I meant that we hear an audio stream typical from an IVR > > application, that says press 1 if you want to . . blabla press two > > if you want to . . . We have checked the configuration and the code > > of our IVR application but we can't see why this is playing, and > > only in some calls, not in all calls. We don't know about the > > procedence of that audio stream . . . Sorry for the double post but > > I thought it wasn't sent. > > Does it sound like *your* IVR? If not, are you using analogue lines > anywhere? > > S > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to sniff RTP and SIP traffic only
Thank you so much ! On Mon, Jun 29, 2009 at 12:21 PM, Duncan Turnbull wrote: > For Linux use tcpdump on the host you are after > > tcpdump udp and port 5060 or portrange 1-16000 -s0 -i eth0 > > where 5060 is your SIP port and 1-16000 are your rtp ranges > -s0 means snap length of 0 so capture all the packet rather than cutting > off at a point > > And refine it by adding the host you are targetting and -w to write to a > file. > > Then you can import the file in wireshark and use the voip utlities to > listen to it fairly easily or use tcpdump -r to read it back and clean > it out a bit more > > Cheers Duncan > > Xavier Cardil wrote: > > Hi, do somebody knows how to sniff RTP and SIP traffic only for a > > faster debugging ? > > > > Thanks. > > > > > > ___ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] unwanted locution
I meant that we hear an audio stream typical from an IVR application, that says press 1 if you want to . . blabla press two if you want to . . . We have checked the configuration and the code of our IVR application but we can't see why this is playing, and only in some calls, not in all calls. We don't know about the procedence of that audio stream . . . Sorry for the double post but I thought it wasn't sent. Thank you. On Mon, Jun 29, 2009 at 11:43 AM, Steve Howes wrote: > Please post ONCE to the list. > > Please define the 'locution'. What words can you hear? (BTW locution > is a rather uncommonly used word in english). > > Steve > > > On 29 Jun 2009, at 10:07, Xavier Cardil wrote: > > > Hi, we are experiencing a problem that is very strange, only on SOME > > calls, a locution jumps in to the RTP stream and both persons > > between the phones can hear it. It is looped and it does not stop > > till hang up. Do you have any clue about what could be happening ? > > > > Thank you ! > > ___ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to sniff RTP and SIP traffic only
Hi, do somebody knows how to sniff RTP and SIP traffic only for a faster debugging ? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] unwanted locution
Hi, we are experiencing a problem that is very strange, only on SOME calls, a locution jumps in to the RTP stream and both persons between the phones can hear it. It is looped and it does not stop till hang up. Do you have any clue about what could be happening ? Thank you ! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] underlying sound during sip calls
Hi we have set up two asterisk machines where we do all the managing of SIP calls. Now, sometimes we call and we get an underlying sound that is a locution from a customer. What could make this to happen ? Is very strange to us, but maybe we are missing something... some configuration, or anything else . . . . Thank you so much. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
