Re: [asterisk-users] SPA-962 Asterisk
On Thu, Nov 06, 2008 at 04:43:07PM -0700, Wilton Helm wrote: The linksys phones annoy me because they cannot implement southern hemisphere DST properly. I was shocked the first time I had to write firmware for an international project. Not only is there the southern hemisphere issue of opposite seasons, but just about anyone in the world with a legislative body has to prove their independence from everyone else by defining the dates a bit differently (not to mention time zones that differ by 15 or 30 minutes). Then the US came along and changed their rules after a million products already had them hard coded in silicon! It's a mess. I just wish we'd all forget about it entirely. Its a way to force people who don't like to get up early to do so anyway. A number of studies have been done on the increase in accidents and reduced worker productivity for a week or two after a change. The recent US change was supposed to save energy, but I suspect if one did a study, they would find that businesses just extended their hours to accommodate a diversity of people, thus increasing their energy consumption! UNIX system supported different timezones with an arbitrary definition ages ago. The timezone only tells the system with what offset to show the time when asked for local time. Sadly some operating systems have this strange concept that changing a time zone means changing the system clock itself. This makes it a huge change indeed. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tired of midget packet received warnings
On Thu, Nov 06, 2008 at 08:42:52AM -0600, Kevin P. Fleming wrote: Louis-David Mitterrand wrote: When monitoring an asterisk through its iax2 port I get these warnings at the console: [Nov 6 13:15:15] WARNING[2209]: chan_iax2.c:7000 socket_process: midget packet received (1 of 4 min) This is triggered by the monitoring app sending a POKE to the iax port. The warning appears even without any '-v'. Your monitoring app is not sending valid IAX2 packets to the server. If it was sending a true IAX2 POKE, it would be a valid packet and wouldn't generate this warning. Could asterisk at least _not_ report this harmless, below-warning event when using a zero-verbose (asterisk -r) level? That would be nice and logical. -- http://www.lesculturelles.net ___ -- 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] tired of midget packet received warnings
Louis-David Mitterrand wrote: On Thu, Nov 06, 2008 at 08:42:52AM -0600, Kevin P. Fleming wrote: Your monitoring app is not sending valid IAX2 packets to the server. If it was sending a true IAX2 POKE, it would be a valid packet and wouldn't generate this warning. Could asterisk at least _not_ report this harmless, below-warning event when using a zero-verbose (asterisk -r) level? That would be nice and logical. Actually, I would have said that corrupt/bad IAX packsets *should* be reported and are *not* harmless. They're harmless in your instance because your monitoring application isn't functioning properly, but to anyone else they're likely to indicate either (a) a hacking attempt or (b) a fairly serious network problem. How about you fix your monitoring application to send a correct IAX2 POKE request? ___ -- 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 - avaya ip office SIP trunking
Hi * Users, I ran into a problem when I was trying to communicate an avaya IP Office talk to asterisk with SIP Trunking. I had successful calls from asterisk to Avaya but not from avaya to asterisk. Can someone provide me insight on how to address it or the path to resolve it. The error I get is mentioned below: (dialing 32564 from avaya to asterisk) [Nov 6 17:14:23] WARNING[6227]: chan_sip.c:8686 get_destination: Huh? Not a SIP header (Tel:+32564 tel:+32564)? [Nov 6 17:14:23] NOTICE[6227]: chan_sip.c:13774 handle_request_invite: Call from 'avayanew' to extension 'Tel:+32564' rejected because extension not found. A SIP Debug of the packet when this happens on asterisk CLI is --- SIP read from 10.10.8.2:5060 --- ACK Tel:+32564 tel:+32564 SIP/2.0 Via: SIP/2.0/UDP 10.10.8.2:5060 ;rport;branch=z9hG4bKb8f50a43f8fce87fda53573e96e498a9 From: avayanew sip:[EMAIL PROTECTED];tag=d60c0430c7b26cbd To: Tel:+32564;tag=as51355066 Call-ID: [EMAIL PROTECTED] CSeq: 152795667 ACK Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO Content-Length: 0 Note: 10.10.8.2 is avaya and 10.10.8.1 is asterisk As I understand, we are getting a Tel URI and a + like in e.164 format and then the number dialed (32564)from avaya. These errors are coming on asterisk console when I try to dial a call from Avaya IP Phone over its SIP trunk on to the asterisk. We probably have to strip the 'Tel:+', so that the asterisk gets the number and thus follows the dialplan programmed in extensions file. Please advise. Any help is appreciated. Thanks as always Regards Krishna ___ -- 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] tired of midget packet received warnings
On 7 Nov 2008, at 08:49, Louis-David Mitterrand wrote: On Thu, Nov 06, 2008 at 08:42:52AM -0600, Kevin P. Fleming wrote: Louis-David Mitterrand wrote: When monitoring an asterisk through its iax2 port I get these warnings at the console: [Nov 6 13:15:15] WARNING[2209]: chan_iax2.c:7000 socket_process: midget packet received (1 of 4 min) This is triggered by the monitoring app sending a POKE to the iax port. The warning appears even without any '-v'. Your monitoring app is not sending valid IAX2 packets to the server. If it was sending a true IAX2 POKE, it would be a valid packet and wouldn't generate this warning. Could asterisk at least _not_ report this harmless, below-warning event when using a zero-verbose (asterisk -r) level? That would be nice and logical. I'd take this warning seriously. It means that your monitoring app isn't monitoring what you think it is. I always want to know when I get malformed protocol packets in. It is always bad news, mostly either a misconfiguration (your case), an attack, (ie my firewall is not protecting this service) or a sign of a switch port going bad. Fix the cause not the symptom. T. ___ -- 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] RFC: multiple packages editing asterisk config files
On Thu, Nov 06, 2008 at 12:57:15PM +0200, Tzafrir Cohen wrote: Hi I'm lately bothered with the need to provide a set of Asterisk configuration files in a package that will be good for a wide range of Asterisk users. Asterisk configuration files support #include and a number of other interesting tricks, as mentioned in http://svn.digium.com/svn/asterisk/branches/1.4/doc/configuration.txt [0]. Let's look at asterisk.conf . Unlike most configuration files, asterisk.conf is not read on reload . Only at startup and restart. It is read by the main asterisk thread and variables from there are globals that may affect all parts of Asterisk. The [directories] section is most useful for anybody who uses a custom installation of Asterisk. Though for distributors I would recommend to patch those values in the source. It is still very handy if you want to have more than one copy of Asterisk on the system. The [options] section has grown over time and includes many options. Some of them corespond to command-line switches. It is interesting to note that values in the configuration file override Asterisk command-line switches of Asterisk and not vice-versa as it is the common with Unix programs. Thus relying on setting parameters through controlling the command-line of Asterisk is not as robust as editing asterisk.conf . The reason for that is that on 'reastart', asterisk re-execs itself. It retains the same command-line options. But it re-reads asterisk.conf , and thus changes to the command-line options would require stopping the Asterisk process and starting it again, as opposed to using the 'restart' command in asterisk. The [compat] section is new in the game for asterisk 1.6 . It makes it actively danbgerous to write asterisk from scratch (as done by, well, someone, in http://svn.digium.com/svn/asterisk/trunk/contrib/scripts/live_ast ). In fact, the only way blessed by the Asterisk developers to write a valid asterisk.conf is to use the output of 'make install' . -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ODBCExec and Asterisk 1.6 New Thread
Thanks, I also ported my app to 1.6.
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Tilghman
Lesher
Enviado el: Friday, November 07, 2008 2:51 AM
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users] ODBCExec and Asterisk 1.6 New Thread
On Thursday 06 November 2008 18:59:29 Sebastian Gutierrez wrote:
Dou you have any example? Can I call directly to querys without the
templates???
func_odbc.conf:
[EXEC]
read=${ARG1}
write=${ARG1}
dsn=something
extensions.conf:
exten = 123,1,Set(result=${ODBC_EXEC(SELECT foo FROM bar)})
--
Tilghman
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[asterisk-users] is it possible to deactivate RTCP?
Hi! Is it possible to deactivate RTCP? (I am using 1.6) thanks klaus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help with asterisk and avaya SIP trunking
Hi * Users, I ran into a problem when I was trying to communicate an avaya IP Office talk to asterisk with SIP Trunking. I had successful calls from asterisk to Avaya but not from avaya to asterisk. Can someone provide me insight on how to address it or the path to resolve it. The error I get is mentioned below: (dialing 32564 from avaya to asterisk) [Nov 6 17:14:23] WARNING[6227]: chan_sip.c:8686 get_destination: Huh? Not a SIP header (Tel:+32564)? [Nov 6 17:14:23] NOTICE[6227]: chan_sip.c:13774 handle_request_invite: Call from 'avayanew' to extension 'Tel:+32564' rejected because extension not found. A SIP Debug of the packet when this happens on asterisk CLI is --- SIP read from 10.10.8.2:5060 --- ACK Tel:+32564 SIP/2.0 Via: SIP/2.0/UDP 10.10.8.2:5060 ;rport;branch=z9hG4bKb8f50a43f8fce87fda53573e96e498a9 From: avayanew sip:[EMAIL PROTECTED];tag=d60c0430c7b26cbd To: Tel:+32564;tag=as51355066 Call-ID: [EMAIL PROTECTED] CSeq: 152795667 ACK Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO Content-Length: 0 Note: 10.10.8.2 is avaya and 10.10.8.1 is asterisk As I understand, we are getting a Tel URI and a + like in e.164 format and then the number dialed (32564)from avaya. These errors are coming on asterisk console when I try to dial a call from Avaya IP Phone over its SIP trunk on to the asterisk. We probably have to strip the 'Tel:+', so that the asterisk gets the number and thus follows the dialplan programmed in extensions file. Please advise. Any help is appreciated. Thanks as always Regards Krishna ___ -- 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] T.38 without port changes
Hi! For T.38 Asterisk uses the port defined in udptl.conf. Is there a workaround (I am using 1.6) for using the same port as RTP also for UDPTL? regards klaus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 776M Any good for connection to local Asterisk server?
I'm pretty sure it is a VERY OLD ISDN router and you can't use it with *. The voice ports have no VoIP capabilities, they are just used directly from the ISDN line. Ronny Julian wrote: I found this at a local sale. I need to find a power supply for it. Before I do I wonder if anyone can tell me if it is any good for Asterisk? Looks to have 4 Ethernet ports and two phone ports. I did get the Cisco serial cable and some documentation. Also will this work with most any Cisco power supply? I see they all share the connector. Thanks! Ronny Julian ___ -- 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] 1.6 Production ready??
Anyone is using 1.6 in production?? Is it ready? ___ -- 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] AEL NoOp not working [SOLVED]
2008/11/6 Steve Murphy [EMAIL PROTECTED] On Thu, 2008-11-06 at 13:55 +0100, Olivier wrote: Yes, you're right : NoOp needs verbosity of 3 and above. Thanks for helping. The surprising thing is that AEL Verbose prints output whatever the verbosity level is (even with 0). Would you qualify this as normal ? Olivier-- The Verbose() app behaves the same whether you call it from AEL or via extensions.conf, or any other method that is used to get dialplan stuff into Asterisk. Are you including the verbosity level? For instance, if you say Verbose(Hi there); the verbosity level is zero by default. If you want to restrict it 3 or more, then Verbose(3,Hello); should do the trick. murf Hi, Ok : I didn't know that I should have read doc more deeply before asking here Thanks -- Steve Murphy Software Developer Digium ___ -- 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] TE121B Doesn't Fit PCI-E Slot
I have 2 HP Proliant 365 G5 servers with PCI-E risers. I bought a Digium TE121B single port card. When installing the card, the slot on the card doesn't quite line up with the tab in the PCI-E slot. If I loosen the front plate on the card, Ican sort of make it plug in, however, the card won't go in far enough to screw down the plate. I tried the card in the other server and had the same problem. Has anyone else experienced this? ___ -- 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] help with dialplan
I have a small system, server, client and 2 phones. Phones are polycom 501's. In general all is working fine. I can call the two phones, speak etc... I can have the server call each phone and play a wave file. However, when trying to setup a direct dial number of 1044 that calls another machine running asterisk - something ODD is happening. ; This is not working [smvoice-sip] exten = 1044,1,Dial(SIP/devcentos5x64_to_bt610tmm/1044) exten = 1044,n,Hangup ; changing 1044 to 10 works find. [smvoice-sip] exten = 10,1,Dial(SIP/devcentos5x64_to_bt610tmm/1044) exten = 10,n,Hangup I am running 1.4.22 and DAHDI 2.0.0 complete. Why is it picking up 10 when trying to dial 1044. How can I determine what is going on here. Thanks, Jerry This is the SIP debug for the 1044 case that does not work. - Use 'exit' when done Asterisk 1.4.22, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer [EMAIL PROTECTED] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. = == Parsing '/etc/asterisk/asterisk.conf': Found [0;37;40m[1;30;40m == [0;37;40mParsing '/etc/asterisk/extconfig.conf': Found [0mConnected to Asterisk 1.4.22 currently running on devcentos5x64 (pid = 3127) devcentos5x64*CLI Verbosity is at least 5 [Kdevcentos5x64*CLI --- SIP read from 192.168.1.89:5060 --- INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.89;branch=z9hG4bKb872214aDBECCC5D From: 404 sip:[EMAIL PROTECTED];tag=25AB8538-7BACFE71 To: sip:[EMAIL PROTECTED];user=phone CSeq: 1 INVITE Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.0.0258 Supported: 1?00rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 249 v=0 o=- 1226069152 1226069152 IN IP4 192.168.1.89 s=Polycom IP Phone c=IN IP4 192.168.1.89 t=0 0 m=audio 2244 RTP/AVP 0 8 18 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 - --- (14 headers 11 lines) --- Sending to 192.168.1.89 : 5060 (no NAT) Using INVITE request as basis request - [EMAIL PROTECTED] --- Reliably Transmitting (no NAT) to 192.168.1.89:5060 --- SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.89;branch=z9hG4bKb872214aDBECCC5D;received=192.168.1.89 From: 404 sip:[EMAIL PROTECTED];tag=25AB8538-7BACFE71 To: sip:[EMAIL PROTECTED];user=phone;tag=as5a3d998e Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces? Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=1f1b706f Content-Length: 0 Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: INVITE) Found user '404' ?? --- SIP read from 192.168.1.89:5060 --- ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.89;branch=z9hG4bKb872214aDBECCC5D From: 404 sip:[EMAIL PROTECTED];tag=25AB8538-7BACFE71 To: sip:[EMAIL PROTECTED];user=phone;tag=as5a3d998e CSeq: 1 ACK Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.0.0258 Max-Forwards: ?70 Content-Length: 0 - --- (11 headers 0 lines) --- ? [Kdevcentos5x64*CLI --- SIP read from 192.168.1.89:5060 --- INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.89;branch=z9hG4bK9c456f3360D09552 From: 404 sip:[EMAIL PROTECTED];tag=25AB8538-7BACFE71 To: sip:[EMAIL PROTECTED];user=phone CSeq: 2 INVITE Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.0.0258 Supported: 1?00rel,replaces Allow-Events: talk,hold,conference Proxy-Authorization: Digest username=404, realm=asterisk, nonce=1f1b706f, uri=sip:[EMAIL PROTECTED];user=phone, response=c6e14f94fa0bbe3d742b6f570982ed79, algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 249 v=0 o=- 1226069152 1226069152 IN IP4 192.168.1.89 s=Polycom IP Phone c=IN IP4 192.168.1.89 t=0 0 m=audio 2244 RTP/AVP 0 8 18 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 - --- (15 headers 11 lines) --- Sending to 192.168.1.89 : 5060 (no NAT) Using INVITE request as basis request - [EMAIL PROTECTED] Found user '404' Found RTP audio
Re: [asterisk-users] tired of midget packet received warnings
On 7 Nov 2008, at 09:57, Louis-David Mitterrand wrote: On Fri, Nov 07, 2008 at 09:29:20AM +, Tim Panton wrote: Your monitoring app is not sending valid IAX2 packets to the server. If it was sending a true IAX2 POKE, it would be a valid packet and wouldn't generate this warning. Could asterisk at least _not_ report this harmless, below-warning event when using a zero-verbose (asterisk -r) level? That would be nice and logical. I'd take this warning seriously. It means that your monitoring app isn't monitoring what you think it is. Granted, the monitoring app is simple minded: it only checks if a port is open. In that respect is does a hell of a good job: I hear a beeping alarm as soon as an asterisk instance goes south. Yep, but it won't tell you that the single IAX thread is blocked in a database access, so asterisk is ignoring your packets, it just hasn't closed the port. So what you are saying is that all monitoring apps should speak native iax, else they are bad? Simply checking if a port is open means it's misconfigured or badly written? I wouldn't go so far. Small generic port-monitoring apps should be allowed to check on asterisk without raising such spurious warnings. You know what happens when crying wolf to often, no one listens after a while. A midget packet is not corrupted, I do have a stateful firewall (fiaif) to intercept those. Kinda, certainly I'd be inclined to write a little plug-in that sends a valid POKE packet. Tell me what your monitor supports and I'll help you craft a valid packet. rant AFAIK the onus is on asterisk to adapat: I've suffered too long of the infamous iax2 port-clogging bug that would and render a server 'unreachable' for no good reason. So much so that I went off iax2 entirely and use SIP exclusively for inter-asterisk communication. So much for the muched touted new and advanced pbx communication protocol the iax2 was sold for! This deal-breaker bug went unfixed for years until recently, despite numerous asterisk users reporting iax2 anomalies month after month. A I bitter? yes. Do I trust Digium folks to know their stuff about what is correct or not in networking protocols? I'll let you guess the answer. /rant Yeah, that one took _way_ too long to fix, I think the problem was that IAX was undocumented so not many people could fix it, that and the fact that it required a major re-code to get chan_iax2 multithreaded. Ed Guy et al have done loads of work on the RFC, to the point where it is actually possible to implement IAX without looking at the asterisk code :-) so the situation is better now. I always want to know when I get malformed protocol packets in. It is always bad news, mostly either a misconfiguration (your case), an attack, (ie my firewall is not protecting this service) or a sign of a switch port going bad. Fix the cause not the symptom. 'fraid I stand by that bit Tim. ___ -- 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] help with dialplan
Jerry Geis wrote: I have a small system, server, client and 2 phones. Phones are polycom 501's. In general all is working fine. I can call the two phones, speak etc... I can have the server call each phone and play a wave file. However, when trying to setup a direct dial number of 1044 that calls another machine running asterisk - something ODD is happening. ; This is not working [smvoice-sip] exten = 1044,1,Dial(SIP/devcentos5x64_to_bt610tmm/1044) exten = 1044,n,Hangup ; changing 1044 to 10 works find. [smvoice-sip] exten = 10,1,Dial(SIP/devcentos5x64_to_bt610tmm/1044) exten = 10,n,Hangup I am running 1.4.22 and DAHDI 2.0.0 complete. Why is it picking up 10 when trying to dial 1044. How can I determine what is going on here. Thanks, Jerry debug snipped Are your polycom phones set up for overlap dialing or do you dial the number then press a key to dial? ___ -- 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] help with dialplan
Are your polycom phones set up for overlap dialing or do you dial the number then press a key to dial? From you message I tried a couple things... Clicking New call, then starting to dial this is when it messes up. when I start entering the number first then click dial this successfull does the 1044 and I am connected as I thought. How do I turn off this overlap dial? Thanks so much. jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help with dialplan
Jerry Geis wrote: How do I turn off this overlap dial? You need to review the dialing rules for the Polycoms. They'd be located in the ftp directory that you've setup for your Polycoms to pull their configs from. It's located in the sip.cfg. Look for the line: digitmap dialplan.digitmap= Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DNS A queries for channel
Hi folks, I've been using * for quite a few years and everyday it surprises me more. I was recently analysing some captures with ethereal/wireshark and found out that * was doing DNS A queries for domain names like channel.mydomain.comwhere channel is the typical string of the dstchannel or channel field in the CDR entries. Obviously those queries returned with negative answer because it does not exists such domainname. My question is why is * asking the DNS for the A entry of the channel? It looks like it does the DNS query upon receiving a SIP message but none SIP header contains the channel string in the SIP headers so it must be something internal, maybe some end-point check? Considering how delicate is * to DNS failures I would like to know whether this behaviour can be disabled in the config files because it makes * block easier and charges the DNS server of senseless queries. I don't know about * internals so it 's far beyond my knowledge following the reception and treatment of SIP message throughout the sip_channel.c code so I would really appreciate any hint about this issue. The capture was done on a 1.4.18 version but I've checked same behaviour (ngrep port 53) on other 1.4 and 1.2 installations. Does anyone knows if this has changed in 1.6? Any help would be really appreciated. Thanks, Samuel. ___ -- 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] ISDN Cause Code 100, Bosch Integral Management Connection
Wolfgang Pichler wrote:
Hi all,
we have the following setup
PSTN 3 PRI Lines --- Asterisk (1.4.22) --- Siemens HiCom
--- Bosch Integral
The Asterisk Machine does play the man in the middle - and adds some
extra functionality to the system (SIP users...) - the normal calls are
getting 1:1 through the system (incoming calls from PSTN are handled by
a simple Dial(ZAP/g1/${EXTEN}) (g1 = Siemens side) - so no special
handling here...
Everything is working as it should - beside of one little thing. The
Bosch Integral PBX does have a special extension (99) which is used to
remote manage the machine - this managment connection is working fine
without asterisk, as soon as asterisk is connected in the middle the
management connection wont work any more - getting back isdn cause code
100. I have already tried dial options d und c (make it digital - clear
channel) - no success.
Can you also post the incoming setup message to your asterisk system ?
They should be almost identical.
Best regards
Hans
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Re: [asterisk-users] help with dialplan
On Fri, Nov 7, 2008 at 11:20 AM, Brent Davidson [EMAIL PROTECTED] wrote: Jerry Geis wrote: Are your polycom phones set up for overlap dialing or do you dial the number then press a key to dial? From you message I tried a couple things... Clicking New call, then starting to dial this is when it messes up. when I start entering the number first then click dial this successfull does the 1044 and I am connected as I thought. How do I turn off this overlap dial? Thanks so much. jerry It's been a while since I've used a polycom so I'm trying to look it up. From what I can see the automatic dialing in Polycoms is accomplished with the digitmap setting. Any of the patterns set in digitmap are dialed automatically as soon as one is recognized. You can try removing everything from the digitmap to force users to click dial on every call. You could do that, or you could read the extensive writeups on www.voip-info.org and figure out a phone dialplan that works for you. That would be my long term suggestion. I try to replicate a POTS line as much as possible, or at least an office phone, with 9 to get out since most people are already hard wired for that in an office environment. The last thing you need is someone trying to dial 911 or whatever your emergency number is and in panic, forgetting to press dial. It isn't that hard to understand, and I was forced to since different regions have seven digit dialing but it is all ten or eleven in the Maryland area. -- 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
Re: [asterisk-users] help with dialplan
On Fri, Nov 7, 2008 at 11:27 AM, Doug Lytle [EMAIL PROTECTED] wrote: Jerry Geis wrote: How do I turn off this overlap dial? You need to review the dialing rules for the Polycoms. They'd be located in the ftp directory that you've setup for your Polycoms to pull their configs from. It's located in the sip.cfg. Look for the line: digitmap dialplan.digitmap= Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. For two phones, I would just use the web interface.. That is of course if you plan on keeping a small amount of phones. -- 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
Re: [asterisk-users] tired of midget packet received warnings
On Fri, Nov 07, 2008 at 09:29:20AM +, Tim Panton wrote: Your monitoring app is not sending valid IAX2 packets to the server. If it was sending a true IAX2 POKE, it would be a valid packet and wouldn't generate this warning. Could asterisk at least _not_ report this harmless, below-warning event when using a zero-verbose (asterisk -r) level? That would be nice and logical. I'd take this warning seriously. It means that your monitoring app isn't monitoring what you think it is. Granted, the monitoring app is simple minded: it only checks if a port is open. In that respect is does a hell of a good job: I hear a beeping alarm as soon as an asterisk instance goes south. So what you are saying is that all monitoring apps should speak native iax, else they are bad? Simply checking if a port is open means it's misconfigured or badly written? I wouldn't go so far. Small generic port-monitoring apps should be allowed to check on asterisk without raising such spurious warnings. You know what happens when crying wolf to often, no one listens after a while. A midget packet is not corrupted, I do have a stateful firewall (fiaif) to intercept those. rant AFAIK the onus is on asterisk to adapat: I've suffered too long of the infamous iax2 port-clogging bug that would and render a server 'unreachable' for no good reason. So much so that I went off iax2 entirely and use SIP exclusively for inter-asterisk communication. So much for the muched touted new and advanced pbx communication protocol the iax2 was sold for! This deal-breaker bug went unfixed for years until recently, despite numerous asterisk users reporting iax2 anomalies month after month. A I bitter? yes. Do I trust Digium folks to know their stuff about what is correct or not in networking protocols? I'll let you guess the answer. /rant I always want to know when I get malformed protocol packets in. It is always bad news, mostly either a misconfiguration (your case), an attack, (ie my firewall is not protecting this service) or a sign of a switch port going bad. Fix the cause not the symptom. T. ___ -- 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 -- http://www.lesculturelles.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6 Production ready??
Sebastian Gutierrez wrote: Anyone is using 1.6 in production?? Is it ready? I have a number of people using 1.6 in production doing SS7-SIP, SS7-IAX, and SS7-ISDN gatewaying. One example (doing SS7-IAX): System uptime: 10 weeks, 6 days, 21 hours, 35 minutes, 45 seconds 8617029 calls processed --- Matthew Fredrickson Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help with dialplan
Steve Totaro wrote: For two phones, I would just use the web interface.. That is of course if you plan on keeping a small amount of phones. Or, if you absolutely hate the web interface :-P -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is it possible to deactivate RTCP?
So, you don't want any media? No audio, video, just sip packets? If you just want a sip router with no media look into SER. Klaus Darilion wrote: Hi! Is it possible to deactivate RTCP? (I am using 1.6) thanks klaus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6 Production ready??
On Fri, Nov 7, 2008 at 11:50 AM, Matthew Fredrickson [EMAIL PROTECTED]wrote: Sebastian Gutierrez wrote: Anyone is using 1.6 in production?? Is it ready? I have a number of people using 1.6 in production doing SS7-SIP, SS7-IAX, and SS7-ISDN gatewaying. One example (doing SS7-IAX): System uptime: 10 weeks, 6 days, 21 hours, 35 minutes, 45 seconds 8617029 calls processed --- Matthew Fredrickson Digium, Inc. EEEK IAX!! Do you use IAX for a reason? Is it because Asterisk does not setup SIP calls very well? Just curious. Impressive, but very purpose specific. Do you only load a couple of modules? I think the question was more along the lines of what Asterisk was meant to be, a feature rich PBX. So let me ask, is anyone using 1.6.x in a production PBX environment with a good amount of features? If so, what has your experience been? If not, did anyone try it and have to resort to the Roll Back Plan? -- 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
[asterisk-users] Providing Ringback
Hello, We've had this problem happen twice with retail customers already and still have no solution. Basically there are times when customers can't get any ring at all. It happens that they call our switch and even though we are receiving ring from the carrier they hear no ring. We have even put a fake-ring(with Rr) back at their request and they are unable to get this ring either. The first time it happened was with a customer running a Cisco switch, now more recently we have a customer with VoipSwitch that gets no ring. Our other customers receive the ring from the carrier fine. Has anyone experienced this before and if so how did you solve it? Regards, Igor Hernandez Escape Communications. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6 Production ready??
Steve Totaro wrote: On Fri, Nov 7, 2008 at 11:50 AM, Matthew Fredrickson [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Sebastian Gutierrez wrote: Anyone is using 1.6 in production?? Is it ready? I have a number of people using 1.6 in production doing SS7-SIP, SS7-IAX, and SS7-ISDN gatewaying. One example (doing SS7-IAX): System uptime: 10 weeks, 6 days, 21 hours, 35 minutes, 45 seconds 8617029 calls processed --- Matthew Fredrickson Digium, Inc. EEEK IAX!! Do you use IAX for a reason? Is it because Asterisk does not setup SIP calls very well? Just curious. The customer chose to use IAX. It has been working very well for him. Impressive, but very purpose specific. Do you only load a couple of modules? Full suite of modules, although it is not using most of them. I did specifically mention in the original message that it was primarily being used as a gateway machine. I think the question was more along the lines of what Asterisk was meant to be, a feature rich PBX. Maybe.. or maybe not. In any case, this is some specific data that someone can use about 1.6's performance. Matthew Fredrickson Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] REFER problems with Asterisk and OpenSER
I've set up an architecture in which OpenSER acts as a registrar and load balancing server for Asterisk machines. I currently have only one Asterisk machine serving as a Media Gateway. My problem is that when A calls B, and then A makes a blind transfer to C, everything works: REFER goes to Asterisk, which processes it, replies with 202 Accepted, and then it generates a valid INVITE to C. Hovewer, when A calls B, and then B attempts a blind call to C, things go awry. REFER from B goes to Asterisk, Asterisk replies with 202 Accepted, and then with 404 Not found in a NOTIFY message. Only difference between these REFER messages is lack of Proxy-Authenticate and Proxy-Authorization headers in the on sent by B. I've read in Asterisk's doxygen doc that all REFER requests have to be properly authenticated. Asterisk in 1.4.21.1 from Debian Lenny, OpenSER is 1.3.2. Any pointers on how to solve this problem? I can give you tcpdump logs of these transactions if required. Regards, Ostrowski Michal ___ -- 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] DNS A queries for channel
On Nov 7, 2008, at 8:29 AM, samuel wrote: Hi folks, I've been using * for quite a few years and everyday it surprises me more. I was recently analysing some captures with ethereal/wireshark and found out that * was doing DNS A queries for domain names like channel.mydomain.com where channel is the typical string of the dstchannel or channel field in the CDR entries. Obviously those queries returned with negative answer because it does not exists such domainname. My question is why is * asking the DNS for the A entry of the channel? It looks like it does the DNS query upon receiving a SIP message but none SIP header contains the channel string in the SIP headers so it must be something internal, maybe some end-point check? Considering how delicate is * to DNS failures I would like to know whether this behaviour can be disabled in the config files because it makes * block easier and charges the DNS server of senseless queries. I don't know about * internals so it 's far beyond my knowledge following the reception and treatment of SIP message throughout the sip_channel.c code so I would really appreciate any hint about this issue. The capture was done on a 1.4.18 version but I've checked same behaviour (ngrep port 53) on other 1.4 and 1.2 installations. Does anyone knows if this has changed in 1.6? Any help would be really appreciated. Thanks, Samuel. That's an interesting discovery, but I suspect it has something to do with a Dial command on a SIP channel. Do you have any idea where in your dialplan these events are occurring? JT --- John Todd [EMAIL PROTECTED]+1-256-428-6083 Asterisk Open Source Community Director ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6 Production ready??
What Hardware? For that performance? -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Matthew Fredrickson Enviado el: Friday, November 07, 2008 3:18 PM Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] 1.6 Production ready?? Steve Totaro wrote: On Fri, Nov 7, 2008 at 11:50 AM, Matthew Fredrickson [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Sebastian Gutierrez wrote: Anyone is using 1.6 in production?? Is it ready? I have a number of people using 1.6 in production doing SS7-SIP, SS7-IAX, and SS7-ISDN gatewaying. One example (doing SS7-IAX): System uptime: 10 weeks, 6 days, 21 hours, 35 minutes, 45 seconds 8617029 calls processed --- Matthew Fredrickson Digium, Inc. EEEK IAX!! Do you use IAX for a reason? Is it because Asterisk does not setup SIP calls very well? Just curious. The customer chose to use IAX. It has been working very well for him. Impressive, but very purpose specific. Do you only load a couple of modules? Full suite of modules, although it is not using most of them. I did specifically mention in the original message that it was primarily being used as a gateway machine. I think the question was more along the lines of what Asterisk was meant to be, a feature rich PBX. Maybe.. or maybe not. In any case, this is some specific data that someone can use about 1.6's performance. Matthew Fredrickson Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is it possible to deactivate RTCP?
I think he wants to leave RTP turned on, but turn off RTCP statistics collection and offers. Sorry I don't have an answer for the actual question, though. Seems reasonable, though perhaps selectable on a per-connection basis. Is RTCP crashing your remote end? JT On Nov 7, 2008, at 9:10 AM, Anthony Francis wrote: So, you don't want any media? No audio, video, just sip packets? If you just want a sip router with no media look into SER. Klaus Darilion wrote: Hi! Is it possible to deactivate RTCP? (I am using 1.6) thanks klaus --- John Todd [EMAIL PROTECTED]+1-256-428-6083 Asterisk Open Source Community Director ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6 Production ready??
I am using it at home with FreePBX on a Clarkconnect 4.3 community server. Only about 10 calls a day but it is doing IAX to SIP. Not had any real problems. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sebastian Gutierrez Sent: Friday, November 07, 2008 11:33 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] 1.6 Production ready?? What Hardware? For that performance? -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Matthew Fredrickson Enviado el: Friday, November 07, 2008 3:18 PM Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] 1.6 Production ready?? Steve Totaro wrote: On Fri, Nov 7, 2008 at 11:50 AM, Matthew Fredrickson [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Sebastian Gutierrez wrote: Anyone is using 1.6 in production?? Is it ready? I have a number of people using 1.6 in production doing SS7-SIP, SS7-IAX, and SS7-ISDN gatewaying. One example (doing SS7-IAX): System uptime: 10 weeks, 6 days, 21 hours, 35 minutes, 45 seconds 8617029 calls processed --- Matthew Fredrickson Digium, Inc. EEEK IAX!! Do you use IAX for a reason? Is it because Asterisk does not setup SIP calls very well? Just curious. The customer chose to use IAX. It has been working very well for him. Impressive, but very purpose specific. Do you only load a couple of modules? Full suite of modules, although it is not using most of them. I did specifically mention in the original message that it was primarily being used as a gateway machine. I think the question was more along the lines of what Asterisk was meant to be, a feature rich PBX. Maybe.. or maybe not. In any case, this is some specific data that someone can use about 1.6's performance. Matthew Fredrickson Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] help with dialplan
Jerry Geis wrote: Are your polycom phones set up for overlap dialing or do you dial the number then press a key to dial? From you message I tried a couple things... Clicking New call, then starting to dial this is when it messes up. when I start entering the number first then click dial this successfull does the 1044 and I am connected as I thought. How do I turn off this overlap dial? Thanks so much. jerry It's been a while since I've used a polycom so I'm trying to look it up. From what I can see the automatic dialing in Polycoms is accomplished with the digitmap setting. Any of the patterns set in digitmap are dialed automatically as soon as one is recognized. You can try removing everything from the digitmap to force users to click dial on every call. ___ -- 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] Help with asterisk and avaya SIP trunking
Krishna Sumanth Chava wrote: Hi * Users, I ran into a problem when I was trying to communicate an avaya IP Office talk to asterisk with SIP Trunking. I had successful calls from asterisk to Avaya but not from avaya to asterisk. Can someone provide me insight on how to address it or the path to resolve it. The error I get is mentioned below: (dialing 32564 from avaya to asterisk) [Nov 6 17:14:23] WARNING[6227]: chan_sip.c:8686 get_destination: Huh? Not a SIP header (Tel:+32564)? [Nov 6 17:14:23] NOTICE[6227]: chan_sip.c:13774 handle_request_invite: Call from 'avayanew' to extension 'Tel:+32564' rejected because extension not found. A SIP Debug of the packet when this happens on asterisk CLI is --- SIP read from 10.10.8.2:5060 http://10.10.8.2:5060 --- ACK Tel:+32564 SIP/2.0 Via: SIP/2.0/UDP 10.10.8.2:5060;rport;branch=z9hG4bKb8f50a43f8fce87fda53573e96e498a9 From: avayanew sip:[EMAIL PROTECTED];tag=d60c0430c7b26cbd To: Tel:+32564;tag=as51355066 Call-ID: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] CSeq: 152795667 ACK Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO Content-Length: 0 Note: 10.10.8.2 http://10.10.8.2 is avaya and 10.10.8.1 http://10.10.8.1 is asterisk As I understand, we are getting a Tel URI and a + like in e.164 format and then the number dialed (32564)from avaya. These errors are coming on asterisk console when I try to dial a call from Avaya IP Phone over its SIP trunk on to the asterisk. We probably have to strip the 'Tel:+', so that the asterisk gets the number and thus follows the dialplan programmed in extensions file. Please advise. Any help is appreciated. Thanks as always Regards Krishna ___ -- 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 you need to make sure the sip dial command in the ipoffice is set to dial 9n; feature dial code n in system the set the dial delay timer to 4 seconds and the dial delay count to 1 this will allow 4 seconds in between each digit there is a setting on the ipo to change the TEL:+ setting to url setting cannot remember wher it is but it in the sip trunk settings robb ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6 Production ready??
Not as impressive as matthew's ref but just to add to the picture. System uptime: 17 weeks, 7 hours, 30 minutes, 51 seconds 342277 calls processed Asterisk SVN-branch-1.6.0-r117951 built by root @ localhost.localdomain on a i686 running Linux on 2008-05-22 21:13:46 UTC using and old Dell 1750 with a quad E1 (TE410) on a Fedora7 Mainly SS7=SIP ,AGI. Freddi. Steve Totaro wrote: On Fri, Nov 7, 2008 at 11:50 AM, Matthew Fredrickson [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Sebastian Gutierrez wrote: Anyone is using 1.6 in production?? Is it ready? I have a number of people using 1.6 in production doing SS7-SIP, SS7-IAX, and SS7-ISDN gatewaying. One example (doing SS7-IAX): System uptime: 10 weeks, 6 days, 21 hours, 35 minutes, 45 seconds 8617029 calls processed --- Matthew Fredrickson Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tired of midget packet received warnings
Tim Panton [EMAIL PROTECTED] writes: I always want to know when I get malformed protocol packets in. It's easy for an attacker to fill your log drive then. /Benny ___ -- 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] Outgoing SIP calls dropped after 30 seconds.
Specs: Asterisk 1.4.22 running behind a SonicWall (transparent mode) with a public IP address. We have our phone system setup as 172.16.2.x that connect through the SonicWall to Asterisk. Incoming calls work flawlessly and we no longer get one-way audio. We are only using SIP (3 trunks now, instead of 2) and having all 3 in use is not an issue. Problem: Make a call on a Polycom 320 IP phone to any number and (4/5 times) it will drop the call after 30 seconds. I noticed that the little timer that pops up on the LCD on the phone is missing when a call will be dropped. This timer appears when the phone is answered, so I have about 30 seconds to talk to them before the call is just dropped. Known Causes: It's a NAT issue, I know that much, I just don't know how to fix it. SIP debugging shows that it attempts to retransmit packets to my phone and since it can't, it drops it after 30 seconds. Log snippet: -- Executing [EMAIL PROTECTED]:19] Dial(SIP/203-b7a2b558, SIP/bw_outbound/+18005551212|300|) in new stack Audio is at public IP port 11968 Adding codec 0x4 (ulaw) to SDP Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 216.82.224.202:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP public IP:5060;branch=z9hG4bK6ea30a1a;rport From: 8881231234 sip:[EMAIL PROTECTED] IP;tag=as3ed791f3 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] IP Call-ID: [EMAIL PROTECTED] IP CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 07 Nov 2008 19:06:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 291 v=0 o=root 21520 21520 IN IP4 151.196.61.115 s=session c=IN IP4 public IP t=0 0 m=audio 11968 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv -- Called bw_outbound/+18885551212 FreePBX*CLI --- SIP read from 216.82.224.202:5060 --- SIP/2.0 100 Giving a try Via: SIP/2.0/UDP public IP:5060;branch=z9hG4bK6ea30a1a;rport=5060 From: 8881231234 sip:[EMAIL PROTECTED] IP;tag=as3ed791f3 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] IP CSeq: 102 INVITE Server: Bandwidth.com TRM (bw7.gold.13) Content-Length: 0 - --- (8 headers 0 lines) --- FreePBX*CLI --- SIP read from 216.82.224.202:5060 --- SIP/2.0 183 Session Progress Via: SIP/2.0/UDP public IP:5060;branch=z9hG4bK6ea30a1a;rport=5060 Record-Route: sip:216.82.224.202;lr;ftag=as3ed791f3 From: 8881231234 sip:[EMAIL PROTECTED] IP;tag=as3ed791f3 To: sip:[EMAIL PROTECTED];tag=VPST50603522629853 Call-ID: [EMAIL PROTECTED] IP CSeq: 102 INVITE Contact: sip:[EMAIL PROTECTED]:5060;transport=udp Content-Type: application/sdp Content-Length: 184 v=0 o=- 1226084867 1226084868 IN IP4 209.244.42.253 s=- c=IN IP4 209.244.42.253 t=0 0 m=audio 64706 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 - --- (10 headers 9 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 209.244.42.253:64706 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 209.244.42.253:64706 -- SIP/bw_outbound-08bf43d0 is making progress passing it to SIP/203-b7a2b558 Audio is at public IP port 16244 Adding codec 0x4 (ulaw) to SDP Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP --- Transmitting (NAT) to 172.16.2.203:5060 --- SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 172.16.2.203;branch=z9hG4bKed293df65EAFD78F;received=172.16.2.203 From: Me sip:[EMAIL PROTECTED] IP;tag=28354B-27A53F00 To: sip:[EMAIL PROTECTED] IP;user=phone;tag=as600b952c Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:[EMAIL PROTECTED] IP Content-Type: application/sdp Content-Length: 291 v=0 o=root 21520 21520 IN IP4 public IP s=session c=IN IP4 public IP t=0 0 m=audio 16244 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv - --- (10 headers 9 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 209.244.42.253:64706 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Re: [asterisk-users] SPA-962 Asterisk
The timezone only tells the system with what offset to show the time when asked for local time. Sadly some operating systems have this strange concept that changing a time zone means changing the system clock itself. This makes it a huge change indeed. Agreed. The firmware I design works the same way--everything internal is in UTC. Any application that must deal with multiple time zones by virtue of market distribution or because it shares time over a network, etc. should use UTC internally and only translate to local time. Using a scheme such as *nix does of an integer rather than broken down field makes the translation trivial. The hard part is deciding how to determine the translation, whether to use hard coded rules, intelligent observation or manual setup. Wilton ___ -- 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] tired of midget packet received warnings
On Fri, Nov 07, 2008 at 09:29:20AM +, Tim Panton wrote: I'd take this warning seriously. It means that your monitoring app isn't monitoring what you think it is. I always want to know when I get malformed protocol packets in. It is always bad news, mostly either a misconfiguration (your case), an attack, (ie my firewall is not protecting this service) or a sign of a switch port going bad. Fix the cause not the symptom. Maybe it's me, but I think that warning should be regarding a problem I can fix. Malformed network content does not neceserily fall under that definition. notice? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outgoing SIP calls dropped after 30 seconds.
Kurt Knudsen wrote: Specs: Asterisk 1.4.22 running behind a SonicWall (transparent mode) with a public IP address. We have our phone system setup as 172.16.2.x that connect through the SonicWall to Asterisk. Incoming calls work flawlessly and we no longer get one-way audio. We are only using SIP (3 trunks now, instead of 2) and having all 3 in use is not an issue. Problem: Make a call on a Polycom 320 IP phone to any number and (4/5 times) it will drop the call after 30 seconds. I noticed that the little timer that pops up on the LCD on the phone is missing when a call will be dropped. This timer appears when the phone is answered, so I have about 30 seconds to talk to them before the call is just dropped. Known Causes: It's a NAT issue, I know that much, I just don't know how to fix it. SIP debugging shows that it attempts to retransmit packets to my phone and since it can't, it drops it after 30 seconds. Log snippet: -- Executing [EMAIL PROTECTED]:19] Dial(SIP/203-b7a2b558, SIP/bw_outbound/+18005551212|300|) in new stack Audio is at public IP port 11968 Adding codec 0x4 (ulaw) to SDP Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 216.82.224.202:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP public IP:5060;branch=z9hG4bK6ea30a1a;rport From: 8881231234 sip:[EMAIL PROTECTED] IP;tag=as3ed791f3 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] IP Call-ID: [EMAIL PROTECTED] IP CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 07 Nov 2008 19:06:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 291 v=0 o=root 21520 21520 IN IP4 151.196.61.115 s=session c=IN IP4 public IP t=0 0 m=audio 11968 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv -- Called bw_outbound/+18885551212 FreePBX*CLI --- SIP read from 216.82.224.202:5060 --- SIP/2.0 100 Giving a try Via: SIP/2.0/UDP public IP:5060;branch=z9hG4bK6ea30a1a;rport=5060 From: 8881231234 sip:[EMAIL PROTECTED] IP;tag=as3ed791f3 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] IP CSeq: 102 INVITE Server: Bandwidth.com TRM (bw7.gold.13) Content-Length: 0 - --- (8 headers 0 lines) --- FreePBX*CLI --- SIP read from 216.82.224.202:5060 --- SIP/2.0 183 Session Progress Via: SIP/2.0/UDP public IP:5060;branch=z9hG4bK6ea30a1a;rport=5060 Record-Route: sip:216.82.224.202;lr;ftag=as3ed791f3 From: 8881231234 sip:[EMAIL PROTECTED] IP;tag=as3ed791f3 To: sip:[EMAIL PROTECTED];tag=VPST50603522629853 Call-ID: [EMAIL PROTECTED] IP CSeq: 102 INVITE Contact: sip:[EMAIL PROTECTED]:5060;transport=udp Content-Type: application/sdp Content-Length: 184 v=0 o=- 1226084867 1226084868 IN IP4 209.244.42.253 s=- c=IN IP4 209.244.42.253 t=0 0 m=audio 64706 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 - --- (10 headers 9 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 209.244.42.253:64706 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 209.244.42.253:64706 -- SIP/bw_outbound-08bf43d0 is making progress passing it to SIP/203-b7a2b558 Audio is at public IP port 16244 Adding codec 0x4 (ulaw) to SDP Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP --- Transmitting (NAT) to 172.16.2.203:5060 --- SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 172.16.2.203;branch=z9hG4bKed293df65EAFD78F;received=172.16.2.203 From: Me sip:[EMAIL PROTECTED] IP;tag=28354B-27A53F00 To: sip:[EMAIL PROTECTED] IP;user=phone;tag=as600b952c Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:[EMAIL PROTECTED] IP Content-Type: application/sdp Content-Length: 291 v=0 o=root 21520 21520 IN IP4 public IP s=session c=IN IP4 public IP t=0 0 m=audio 16244 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv - --- (10 headers 9 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 209.244.42.253:64706 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x4
Re: [asterisk-users] SPA-962 Asterisk
On Fri, Nov 07, 2008 at 12:43:47PM -0700, Wilton Helm wrote: The timezone only tells the system with what offset to show the time when asked for local time. Sadly some operating systems have this strange concept that changing a time zone means changing the system clock itself. This makes it a huge change indeed. Agreed. The firmware I design works the same way--everything internal is in UTC. Any application that must deal with multiple time zones by virtue of market distribution or because it shares time over a network, etc. should use UTC internally and only translate to local time. Using a scheme such as *nix does of an integer rather than broken down field makes the translation trivial. The hard part is deciding how to determine the translation, whether to use hard coded rules, intelligent observation or manual setup. The size of the whole glibc time zones distribution, if you ignore duplicates, is: $ du -sh /usr/share/zoneinfo \ --exclude=/usr/share/zoneinfo/posix \ --exclude=/usr/share/zoneinfo/right 2.3M/usr/share/zoneinfo This includes 577 files. I'm sure you can trim that down. The point is that you then ask the user for the time zone, and don't need the DST checkbox. The device will go start using DST automatically. $ zdump -v /usr/share/zoneinfo/America/New_York | grep 2008 /usr/share/zoneinfo/America/New_York Sun Mar 9 06:59:59 2008 UTC = Sun Mar 9 01:59:59 2008 EST isdst=0 gmtoff=-18000 /usr/share/zoneinfo/America/New_York Sun Mar 9 07:00:00 2008 UTC = Sun Mar 9 03:00:00 2008 EDT isdst=1 gmtoff=-14400 /usr/share/zoneinfo/America/New_York Sun Nov 2 05:59:59 2008 UTC = Sun Nov 2 01:59:59 2008 EDT isdst=1 gmtoff=-14400 /usr/share/zoneinfo/America/New_York Sun Nov 2 06:00:00 2008 UTC = Sun Nov 2 01:00:00 2008 EST isdst=0 gmtoff=-18000 Thus the time zone is not GMT+3 or GMT-5. This the current time zone. But it forces the user to actively change the timezone whenever the DSP come into effect. The time zone is USA/Eastern, Peru, or whatever. And what if those definitions keep changing? e.g. if you live in Brazil? I figure some sort of manual override, such as the explicit GMT[+-]NN zones. And this interface would not be complete without a clock showing the local time according to those settings, I guess. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outgoing SIP calls dropped after 30 seconds.
At 14:15 11/7/2008, SIP wrote: Kurt Knudsen wrote: Specs: Asterisk 1.4.22 running behind a SonicWall (transparent mode) with a public IP address. We have our phone system setup as 172.16.2.x that connect through the SonicWall to Asterisk. Incoming calls work flawlessly and we no longer get one-way audio. We are only using SIP (3 trunks now, instead of 2) and having all 3 in use is not an issue. Question: Why does it sometimes work and sometimes not? This makes no sense and it happens on all phones. Any suggestions? We see this on occasion. It sounds a lot like Asterisk doing its usual routine of deciding that you can't POSSIBLY have a call going through because it can't receive an ACK response properly. Asterisk tries several times to send an ACK and get a response. If the remote system routes ACKs differently than it routes everything else, often times those ACKs get lost, and Asterisk assumes that the call can't be working, so it destroys it. ACK handling is a bit tricky in the real world, and we've run across countless incorrectly-configured SIP servers that don't handle it properly, so calls to them last just about exactly 30 seconds and then drop. There is, unfortunately, no way to turn off Asterisk's 'intelligent' behaviour in this scenario short of possibly patching the code. http://lists.digium.com/pipermail/asterisk-users/2007-May/187951.html ___ -- 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] Outgoing SIP calls dropped after 30 seconds.
Ok, recompiling it now with a 1 instead of XMIT_CRITICAL. Will check back to see if it worked. Would be nice if it did :) Thanks, Kurt On Fri, Nov 7, 2008 at 3:38 PM, Doug [EMAIL PROTECTED] wrote: At 14:15 11/7/2008, SIP wrote: Kurt Knudsen wrote: Specs: Asterisk 1.4.22 running behind a SonicWall (transparent mode) with a public IP address. We have our phone system setup as 172.16.2.x that connect through the SonicWall to Asterisk. Incoming calls work flawlessly and we no longer get one-way audio. We are only using SIP (3 trunks now, instead of 2) and having all 3 in use is not an issue. Question: Why does it sometimes work and sometimes not? This makes no sense and it happens on all phones. Any suggestions? We see this on occasion. It sounds a lot like Asterisk doing its usual routine of deciding that you can't POSSIBLY have a call going through because it can't receive an ACK response properly. Asterisk tries several times to send an ACK and get a response. If the remote system routes ACKs differently than it routes everything else, often times those ACKs get lost, and Asterisk assumes that the call can't be working, so it destroys it. ACK handling is a bit tricky in the real world, and we've run across countless incorrectly-configured SIP servers that don't handle it properly, so calls to them last just about exactly 30 seconds and then drop. There is, unfortunately, no way to turn off Asterisk's 'intelligent' behaviour in this scenario short of possibly patching the code. http://lists.digium.com/pipermail/asterisk-users/2007-May/187951.html ___ -- 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] Outgoing SIP calls dropped after 30 seconds.
That seems to have sort of worked. It seems the phone decided to end the call this time, instead of Asterisk and now the call is dangling inside of 'sip show channels'. So that solution didn't work :( On Fri, Nov 7, 2008 at 4:28 PM, Kurt Knudsen [EMAIL PROTECTED] wrote: Ok, recompiling it now with a 1 instead of XMIT_CRITICAL. Will check back to see if it worked. Would be nice if it did :) Thanks, Kurt On Fri, Nov 7, 2008 at 3:38 PM, Doug [EMAIL PROTECTED] wrote: At 14:15 11/7/2008, SIP wrote: Kurt Knudsen wrote: Specs: Asterisk 1.4.22 running behind a SonicWall (transparent mode) with a public IP address. We have our phone system setup as 172.16.2.x that connect through the SonicWall to Asterisk. Incoming calls work flawlessly and we no longer get one-way audio. We are only using SIP (3 trunks now, instead of 2) and having all 3 in use is not an issue. Question: Why does it sometimes work and sometimes not? This makes no sense and it happens on all phones. Any suggestions? We see this on occasion. It sounds a lot like Asterisk doing its usual routine of deciding that you can't POSSIBLY have a call going through because it can't receive an ACK response properly. Asterisk tries several times to send an ACK and get a response. If the remote system routes ACKs differently than it routes everything else, often times those ACKs get lost, and Asterisk assumes that the call can't be working, so it destroys it. ACK handling is a bit tricky in the real world, and we've run across countless incorrectly-configured SIP servers that don't handle it properly, so calls to them last just about exactly 30 seconds and then drop. There is, unfortunately, no way to turn off Asterisk's 'intelligent' behaviour in this scenario short of possibly patching the code. http://lists.digium.com/pipermail/asterisk-users/2007-May/187951.html ___ -- 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] 1.6 Production ready??
Sebastian Gutierrez wrote: What Hardware? For that performance? It's a dual core 1.8 GHz Opteron with 2 TE420P cards and 4 GB of RAM. Oh yeah, those numbers indicate averaging over 110,000 calls per day (the ones I posted below) :-) -- Matthew Fredrickosn Digium, Inc. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Matthew Fredrickson Enviado el: Friday, November 07, 2008 3:18 PM Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] 1.6 Production ready?? Steve Totaro wrote: On Fri, Nov 7, 2008 at 11:50 AM, Matthew Fredrickson [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Sebastian Gutierrez wrote: Anyone is using 1.6 in production?? Is it ready? I have a number of people using 1.6 in production doing SS7-SIP, SS7-IAX, and SS7-ISDN gatewaying. One example (doing SS7-IAX): System uptime: 10 weeks, 6 days, 21 hours, 35 minutes, 45 seconds 8617029 calls processed --- Matthew Fredrickson Digium, Inc. EEEK IAX!! Do you use IAX for a reason? Is it because Asterisk does not setup SIP calls very well? Just curious. The customer chose to use IAX. It has been working very well for him. Impressive, but very purpose specific. Do you only load a couple of modules? Full suite of modules, although it is not using most of them. I did specifically mention in the original message that it was primarily being used as a gateway machine. I think the question was more along the lines of what Asterisk was meant to be, a feature rich PBX. Maybe.. or maybe not. In any case, this is some specific data that someone can use about 1.6's performance. Matthew Fredrickson Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] 1.6 Production ready??
Sebastian Gutierrez wrote: What Hardware? For that performance? It's a dual core 1.8 GHz Opteron with 2 TE420P cards and 4 GB of RAM. Oh yeah, those numbers indicate averaging over 110,000 calls per day (the ones I posted below) :-) -- Matthew Fredrickosn Digium, Inc. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Matthew Fredrickson Enviado el: Friday, November 07, 2008 3:18 PM Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] 1.6 Production ready?? Steve Totaro wrote: On Fri, Nov 7, 2008 at 11:50 AM, Matthew Fredrickson [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Sebastian Gutierrez wrote: Anyone is using 1.6 in production?? Is it ready? I have a number of people using 1.6 in production doing SS7-SIP, SS7-IAX, and SS7-ISDN gatewaying. One example (doing SS7-IAX): System uptime: 10 weeks, 6 days, 21 hours, 35 minutes, 45 seconds 8617029 calls processed --- Matthew Fredrickson Digium, Inc. EEEK IAX!! Do you use IAX for a reason? Is it because Asterisk does not setup SIP calls very well? Just curious. The customer chose to use IAX. It has been working very well for him. Impressive, but very purpose specific. Do you only load a couple of modules? Full suite of modules, although it is not using most of them. I did specifically mention in the original message that it was primarily being used as a gateway machine. I think the question was more along the lines of what Asterisk was meant to be, a feature rich PBX. Maybe.. or maybe not. In any case, this is some specific data that someone can use about 1.6's performance. Matthew Fredrickson Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] tired of midget packet received warnings
Rob Hillis wrote: Louis-David Mitterrand wrote: On Thu, Nov 06, 2008 at 08:42:52AM -0600, Kevin P. Fleming wrote: Your monitoring app is not sending valid IAX2 packets to the server. If it was sending a true IAX2 POKE, it would be a valid packet and wouldn't generate this warning. Could asterisk at least _not_ report this harmless, below-warning event when using a zero-verbose (asterisk -r) level? That would be nice and logical. Actually, I would have said that corrupt/bad IAX packsets *should* be reported and are *not* harmless. They're harmless in your instance because your monitoring application isn't functioning properly, but to anyone else they're likely to indicate either (a) a hacking attempt or (b) a fairly serious network problem. How about you fix your monitoring application to send a correct IAX2 POKE request? Personally, I just like reading the word 'midget' . It makes me smile. PaulH ___ -- 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] TE121B Doesn't Fit PCI-E Slot
I have 2 HP Proliant 365 G5 servers with PCI-E risers. I bought a Digium TE121B single port card. When installing the card, the slot on the card doesn't quite line up with the tab in the PCI-E slot. If I loosen the front plate on the card, Ican sort of make it plug in, however, the card won't go in far enough to screw down the plate. I tried the card in the other server and had the same problem. Has anyone else experienced this? Probably obvious, but did the assembler place the bracket on the wrong side of the pcb forcing the offset? jlc ___ -- 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] tired of midget packet received warnings
Tzafrir Cohen wrote: On Fri, Nov 07, 2008 at 09:29:20AM +, Tim Panton wrote: I'd take this warning seriously. It means that your monitoring app isn't monitoring what you think it is. I always want to know when I get malformed protocol packets in. It is always bad news, mostly either a misconfiguration (your case), an attack, (ie my firewall is not protecting this service) or a sign of a switch port going bad. Fix the cause not the symptom. Maybe it's me, but I think that warning should be regarding a problem I can fix. Malformed network content does not neceserily fall under that definition. notice? Absolutely it does. Warnings of malformed packets are often (as mentioned above) symptomatic of network problems. Fix the network problem, fix the warning. ___ -- 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] Outgoing SIP calls dropped after 30 seconds.
To get to the bottom of it I'd recommend determining why the ACKs are not getting through to Asterisk rather than trying to work around it. I'm actually suprised Asterisk terminates the call by default when it doesn't get the ACK to it's 200 Ok response that must be new for 1.4.22 as I haven't seen that behaviour in earlier versions. In my opinion it's unwarranted behaviour, if Asterisk is getting RTP then it should leave the call up irrespective of whether it gets an ACK or not. From the original SIP trace the ACK does not appear to be arriving at your Asterisk server at all. Try doing a packet trace on the network segment where the calling SIP agent is and see where it's trying to send the ACK to. My guess would be your firewall is incorrectly handling the SIP messages. Generally it's very bad news to use an ALG or firewall to mangle SIP packets as they almost always get it wrong. In your case there is a Record-Route header in the response so the ACK request should be being sent to that address. Perhaps your firewall is not correctly mangling that to allow the request to find its way back to your Asterisk server. Record-Route: sip:216.82.224.202;lr;ftag=as3ed791f3 Regards, Greyman. ___ -- 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] Providing Ringback
Hi Igor,
We had an interconnect with a carrier that generated early media for
progress indications but the carrier's switch, in this case a Cerpack,
would only start sending the RTP for the early media AFTER it received
an RTP packet from the Asterisk end. Completely stupid behaviour since
early media is generally only one way but that's what it did.
We worked around it by recording 200ms of silence and playing that
back to the carrier's Cerpack with the Background command whenever we
received an incoming call. This got two way RTP set up and allowed the
progress tones to be correctly passed through to the user.
[noringback]
exten = _X.,1,Background(/var/lib/asterisk/custom-sounds/silence_200,n)
exten = _X.,2,Goto(incoming, ${EXTEN}, 1)
Regards,
Greyman.
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Re: [asterisk-users] Providing Ringback
Thanks a lot Grey. I'll look into it.
Regards,
--
Igor Hernandez
Escape Communications
http://www.escapetel.com
Grey Man wrote:
Hi Igor,
We had an interconnect with a carrier that generated early media for
progress indications but the carrier's switch, in this case a Cerpack,
would only start sending the RTP for the early media AFTER it received
an RTP packet from the Asterisk end. Completely stupid behaviour since
early media is generally only one way but that's what it did.
We worked around it by recording 200ms of silence and playing that
back to the carrier's Cerpack with the Background command whenever we
received an incoming call. This got two way RTP set up and allowed the
progress tones to be correctly passed through to the user.
[noringback]
exten = _X.,1,Background(/var/lib/asterisk/custom-sounds/silence_200,n)
exten = _X.,2,Goto(incoming, ${EXTEN}, 1)
Regards,
Greyman.
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Re: [asterisk-users] tired of midget packet received warnings
On Sat, Nov 08, 2008 at 02:33:18PM +1100, Rob Hillis wrote: Tzafrir Cohen wrote: On Fri, Nov 07, 2008 at 09:29:20AM +, Tim Panton wrote: I'd take this warning seriously. It means that your monitoring app isn't monitoring what you think it is. I always want to know when I get malformed protocol packets in. It is always bad news, mostly either a misconfiguration (your case), an attack, (ie my firewall is not protecting this service) or a sign of a switch port going bad. Fix the cause not the symptom. Maybe it's me, but I think that warning should be regarding a problem I can fix. Malformed network content does not neceserily fall under that definition. notice? Absolutely it does. Warnings of malformed packets are often (as mentioned above) symptomatic of network problems. Fix the network problem, fix the warning. C'mon, even firewalls give you the option of _not_ logging malformed packets! fiaif does. Else your logfile would be the weak point of your system. And what if you can't fix the source of these packets? And what if friendly peers outside of your realm (likely to iax-call you, so can't block them) sends these packets? There are holes in your logic. So asterisk has to be puritan of the lot? Holier than thou? Pro-life with malformed packets? I see where this is going and I don't like it one bit. -- http://www.lesculturelles.net ___ -- 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
