Re: [asterisk-users] OT Astricon/Digium Beach Ball Mailing
On Sat, Jul 19, 2008 at 5:48 AM, Steve Totaro [EMAIL PROTECTED] wrote: Cool, but I tried to blow up the beach ball and the the seam where the part opens to inflate the ball was not connected to the ball whatsoever, so it went right in the trash. Steve, It is with great sadness that I read your message. Yes, we just had a thread about the Digium Tweaker (tm), the Digium Pen (tm) and the Digium Mouse Pag (tm) and darn it, they've always provided the best swag of any tech company. Now this news. The world has become a sad place indeed! removing tongue from cheek you're right to tell them about this, though, for the reason you stated. Thanks for sharing :) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] OT Astricon/Digium Beach Ball Mailing
Steve Totaro wrote: Just an FYI for Digium. I received a mailing today from you guys which was nice. The price of mailing was ~$1.60 and inside was an inflatable beach ball. Cool, but I tried to blow up the beach ball and the the seam where the part opens to inflate the ball was not connected to the ball whatsoever, so it went right in the trash. I wonder if the sick heat had anything to do with it, was mine just bad, or should Digium get a refund from the promotion company for providing garbage? Anyone else get one? Was it OK or junk? I post this not to put down Digium, the thought was nice, I wish I could play with my Digium beach ball, but Digium should know about it if it was common. Postage alone was costly. I mean this without a hint of sarcasm or derision toward you or Digium, but: Award for ... most bewildering asterisk-users list post ever! :-) -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4.21.1 SIP failing, requiring reboot
Interestingly enough, I've had my Grandstream suffering from the same problem since I upgraded to 1.4.20, although my config is static rather than realtime. I'd actually written it off to typical Grand-heap-of-$#!+-stream behaviour. :) I didn't say because I wanted my original email to limit itself to facts I was sure of, but I think my SIP problems started with 1.4.20 as well. I'm fairly sure 1.4.19 was solid... going back today. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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
[asterisk-users] Explication for ast_safe_system
Can someone please explain the reason on the following code (in asterisk.c, function ast_safe_system()): /* Close file descriptors and launch system command */ for (x = STDERR_FILENO + 1; x 4096; x++) close(x); Why to close so many descriptors? Thanks in advance Éric ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Explication for ast_safe_system
On Saturday 19 July 2008 06:41:04 Eric Dantie wrote: Can someone please explain the reason on the following code (in asterisk.c, function ast_safe_system()): /* Close file descriptors and launch system command */ for (x = STDERR_FILENO + 1; x 4096; x++) close(x); Why to close so many descriptors? There's no way to know how many file descriptors are actually open, and every file descriptor that is open when a process is forked is duplicated in the new process. Also, if the process being forked is long running (such as an AGI), then there could be side effects to not closing all descriptors in a child process (such as receiving a SIGPIPE when the other end closes). Eventually, we could probably start registering the highest file descriptor to a central function, ensuring that we close all of them. It is admittedly a hack to pick an arbitrary number like this. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] OT Astricon/Digium Beach Ball Mailing
On Sat, 2008-07-19 at 03:40 -0400, Alex Balashov wrote: Steve Totaro wrote: Just an FYI for Digium. I received a mailing today from you guys which was nice. The price of mailing was ~$1.60 and inside was an inflatable beach ball. Cool, but I tried to blow up the beach ball and the the seam where the part opens to inflate the ball was not connected to the ball whatsoever, so it went right in the trash. I wonder if the sick heat had anything to do with it, was mine just bad, or should Digium get a refund from the promotion company for providing garbage? Anyone else get one? Was it OK or junk? I didn't get one? Where do I sign up to receive these balls (preferrably working ones) and pens? I keep buying digium stuff already ;) Conrad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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
[asterisk-users] Beginner Questions part II
I should start with a thank you to the list for helping me getting up and running with Asterisk about a week ago. I have been happily fiddling with Asterisk since then :). I am working on adding a couple features to my dialplan. My setup involves my asterisk box connecting to another third party sip provider. I configured the extra trunk and there are no issues passing calls through their systems. As it stands right now, I setup a calling rule that matches the pattern 9-XXX-XXX-, stripes off the 9, and then passes the call through to the third party. I would like to just dial XXX-XXX- without having to dial the extra 9. Is there a way that I can configure asterisk so that I check to see if the extension exists on my box first, if it does then pickup and if it doesn't then forward the call onto the third party? If so, how? The other feature I am looking to add is *67 caller id blocking. Am I right in thinking that I first would configure an incoming call rule that matches to *67-whatever and then pass unknown as the caller id from there? Any help is greatly appreciated. Even if it is just pointing me in the right direction in regards to reading material. Thanks, John ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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
[asterisk-users] Not a valid SIP contact - Asterisk 1.4.21.1 Mitel SIP phones
Hi, I have a client using Mitel 5212 phones in SIP mode with a 1.4.21.1 Asterisk server (and a couple of previous 1.4 versions). They're mostly happy with the combination except for this one issue. For incoming calls only, either originating from other local SIP phones or from a PRI, calls won't get bridged (remote party get's hung up) if the call is answer too quickly on the Mitel. Or so it seems. The receiving Mitel phone thinks the call is in session though. Oh... this does not happen all of the time, maybe 50%. Asterisk is reporting errors like: [Jul 19 10:46:41] NOTICE[2466]: chan_sip.c:8068 set_address_from_contact: '72.16.1.20;tag=as7b9f4bfb' is not a valid SIP contact (missing sip:) trying to use anyway [Jul 19 10:46:41] WARNING[2466]: chan_sip.c:8097 set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : '72.16.1.20' [Jul 19 10:46:41] WARNING[2466]: chan_sip.c:5839 set_destination: Can't find address for host '72.16.1.20' [Jul 19 10:46:41] WARNING[2466]: chan_sip.c:5839 set_destination: Can't find address for host '72.16.1.20' or [Jul 19 10:45:03] NOTICE[2466]: chan_sip.c:8068 set_address_from_contact: '172.16.1.20;tag=as4a1b11c8' is not a valid SIP contact (missing sip:) trying to use anyway [Jul 19 10:45:03] WARNING[2466]: chan_sip.c:8097 set_address_from_contact: Invalid host name in Contact: can't resolve in DNS) : '172.16.1.20' [Jul 19 10:45:04] WARNING[2466]: chan_sip.c:5839 set_destination: Can't find address for host '172.16.1.20' [Jul 19 10:45:04] WARNING[2466]: chan_sip.c:5839 set_destination: Can't find address for host '172.16.1.20' or [Jul 19 10:52:18] NOTICE[2466]: chan_sip.c:8068 set_address_from_contact: 'nt-Length:0' is not a valid SIP contact (missing sip:) trying to use anyway [Jul 19 10:52:18] WARNING[2466]: chan_sip.c:8097 set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : 'nt-Length' [Jul 19 10:52:18] WARNING[2466]: chan_sip.c:5839 set_destination: Can't find address for host 'nt-Length' [Jul 19 10:52:18] WARNING[2466]: chan_sip.c:5839 set_destination: Can't find address for host 'nt-Length' -- SIP/517-09215fb0 answered SIP/512-09258c78 -- Native bridging SIP/512-09258c78 and SIP/517-09215fb0 [Jul 19 10:52:18] WARNING[22054]: chan_sip.c:5839 set_destination: Can't find address for host 'nt-Length' -- Got SIP response 416 Unsupported URI Scheme back from 172.16.1.157 [Jul 19 10:52:18] WARNING[2466]: chan_sip.c:5839 set_destination: Can't find address for host 'nt-Length' So it seems that the Mitel phone is sending a bad contact field in SIP. I've confirmed via tcpdump that this is what's in the SIP packet on the wire. I wanted to try a different version of SIP on the Mitel but that doesn't seem to be an option, it's not available for download and the local Mitel vendor can't seem to get his hands on anything newer than 6.0.0.something, though there is supposedly 7.1.x available. These phones are running 06.00.00.19. The Asterisk server has a pretty standard sip.conf, bindaddr=0.0.0.0 pedantic=no; bindport=5060 srvlookup=no tos_video=af41 notifyringing=yes notifyhold=yes allowsubscribe=yes limitonpeer=yes localnet=172.16.1.0/255.255.255.0 Polycom phones on this same asterisk server do not display this behavior. I'm wondering if there is a workaround for this apparent Mitel issue in Asterisk's configuration. Anyone using this combination with success? Thanks in advance for any thoughts Mark ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Changinf Polycom-501 config server from remote?
I contacted Polycom support about this a few weeks ago - their answer was that is was not possible. We found that hand-configured phones had to be reset by reset device setting in the Polycom phone menu then Option 66 in a DHCP server would override whatever was hand-configured and allow you to auto-provision. Robert On Fri, Jul 18, 2008 at 10:13 PM, Yehavi Bourvine +972-8-9489444 [EMAIL PROTECTED] wrote: Hello, Our Polycom-501 phones are set to retreive their config for the server by a static configuation defined at the phones (boot servers). Is there any way to change it remotely? I found no relevant field in the internal WEB browser, nor anything in the configuration files (sip.conf and phone1.conf). Thanks! __Yehavi: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Not a valid SIP contact - Asterisk 1.4.21.1 Mitel SIP phones
On July 19, 2008 11:22:08 am Mark Wiater wrote: Hi, I have a client using Mitel 5212 phones in SIP mode with a 1.4.21.1 Asterisk server (and a couple of previous 1.4 versions). They're mostly happy with the combination except for this one issue. For incoming calls only, either originating from other local SIP phones or from a PRI, calls won't get bridged (remote party get's hung up) if the call is answer too quickly on the Mitel. Or so it seems. The receiving Mitel phone thinks the call is in session though. Asterisk is reporting errors like: [Jul 19 10:46:41] NOTICE[2466]: chan_sip.c:8068 set_address_from_contact: '72.16.1.20;tag=as7b9f4bfb' is not a valid SIP contact (missing sip:) trying to use anyway [Jul 19 10:46:41] WARNING[2466]: chan_sip.c:8097 set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : '72.16.1.20' [Jul 19 10:46:41] WARNING[2466]: chan_sip.c:5839 set_destination: Can't find address for host '72.16.1.20' [Jul 19 10:46:41] WARNING[2466]: chan_sip.c:5839 set_destination: Can't find address for host '72.16.1.20' Might want to post a sip debug of one of the sessions from the Mitel phone. -- Matt Watson http://www.mattgwatson.ca ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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
[asterisk-users] Echo Issue
I am being told by the users on a purely sip based setup that when an inbound sip call is first answered, they here an echo on their greeting and then the conversation stabilizes and it works well. Any ideas where to look to start curing this? Thanks! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] OT Astricon/Digium Beach Ball Mailing
I've asked a number of others I know in real life who got the beach balls and all are reported as being fully functional. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] asterisk-users Digest, Vol 48, Issue 56
Hola, Estoy de vacaciones hasta el 1 de Agosto. Para dar soporte sobre la centralita de telefonia: [EMAIL PROTECTED] Perdonen las molestias. Ruth Llaneza Lapausa - Tecnico de VoIP. [EMAIL PROTECTED] Tlf: 902 199 384 Mildmac SA � www.mildmac.es � [EMAIL PROTECTED] C/ Hnos. Garc�a Noblejas 41, 6� planta. 28037 - Madrid Tlf: +34 91 501 33 02 Fax: +34 91 501 57 45 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Changinf Polycom-501 config server from remote?
There are many parameters that cannot be defined via the web GUI. These must be defined in the config files. Uptake of the configs is only done at boot time. I recall that the www.voip-info.org wiki has a Perl script for remote forcing a reboot of Polycom phones. Michael On Sat, 19 Jul 2008 08:27:41 -0700, Robert McNaught wrote: I contacted Polycom support about this a few weeks ago - their answer was that is was not possible. We found that hand-configured phones had to be reset by reset device setting in the Polycom phone menu then Option 66 in a DHCP server would override whatever was hand-configured and allow you to auto-provision. Robert On Fri, Jul 18, 2008 at 10:13 PM, Yehavi Bourvine +972-8-9489444 [EMAIL PROTECTED] wrote: Hello, Our Polycom-501 phones are set to retreive their config for the server by a static configuation defined at the phones (boot servers). Is there any way to change it remotely? I found no relevant field in the internal WEB browser, nor anything in the configuration files (sip.conf and phone1.conf). Thanks! __Yehavi: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 No virus found in this incoming message. Checked by AVG - http://www.avg.com Version: 8.0.138 / Virus Database: 270.5.1/1559 - Release Date: 7/17/2008 6:08 PM -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] asterisk-users Digest, Vol 48, Issue 56
Message: 1 Date: Fri, 18 Jul 2008 20:35:47 -0700 From: Dave Platt [EMAIL PROTECTED] I'm preparing for a client install of * by doing a fresh one in-house. Unlike my earlier installation that runs asterisk as superuser, my current experimental box runs without such privilege. This is causing it to moan that it can't set TOS. I absolutely don't want to install it on the client LAN without this capability. If need be, I'll set the binary to run setuid root. But I'm looking for something more elegant. While googling, I found a suggestion to use iptables mangle rules to set TOS for all packets going out of the box on ports like 5060 and 1:2. Not a bad hack, but indiscriminate and this box will be handling other traffic besides the RTP. I'd like to do better. It is possible for an iptables filter/rule to match packets in the OUTPUT chain based on the UID or GID of the process which created them, if you have the owner module loaded. You should be able to add a rule to the OUTPUT chain of the mangle table which will set the TOS properly for any and all outbound packets generated locally by the non-root user ID which you're using to run Asterisk. I've used LARTC and I'm aware of the capability, but keying on UID did not occur to me. Thank you - it's a good solution. Come to think of it, I think I need to do this myself. I'm using the ultimate Linux traffic conditioning configuration (modified very slightly) to prioritize my system's outbound traffic into multiple queues by TOS, and it's probably mis-queueing the RTP traffic because my Debian install of Asterisk is running under a non-root UID. Glad to be of assistance. I thought of using POSIX access control to enable asterisk to do TOS setting without being root (would this be CAP_NET_RAW?), which sounds perfect, but so far I'm operating with stock ubuntu hardy, and I would like to avoid a kernel build to add this capability. Any other ideas? Seems like iptables -t mangle -A OUTPUT -m owner --uid-owner $ASTERISK would be along the lines of what you want? Mark the packets with the TOS you want... and then consider using the Linux traffic-shaping system to make sure that they really do get transmitted ahead of non-urgent packets: Traffic-shaping in the box would probably be overkill for my purpose because the nature of the routing in this box will limit the contention from this source. I think I just need to have the packets treated well once they hit the local network. But this is also a worthwhile consideration, and probably useful in other circumstances. Again, thanks for the reply - it's right on target and solves my problem nicely. smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] OT Astricon/Digium Beach Ball Mailing
On Sat, Jul 19, 2008 at 7:09 PM, Alex Balashov [EMAIL PROTECTED] wrote: I've asked a number of others I know in real life who got the beach balls and all are reported as being fully functional. So this is not a case for the bug tracker? Perhaps a bounty... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Not a valid SIP contact - Asterisk 1.4.21.1 Mitel SIP phones
Matt Watson wrote: On July 19, 2008 11:22:08 am Mark Wiater wrote: Hi, I have a client using Mitel 5212 phones in SIP mode with a 1.4.21.1 Asterisk server (and a couple of previous 1.4 versions). They're mostly happy with the combination except for this one issue. For incoming calls only, either originating from other local SIP phones or from a PRI, calls won't get bridged (remote party get's hung up) if the call is answer too quickly on the Mitel. Or so it seems. The receiving Mitel phone thinks the call is in session though. Asterisk is reporting errors like: [Jul 19 10:46:41] NOTICE[2466]: chan_sip.c:8068 set_address_from_contact: '72.16.1.20;tag=as7b9f4bfb' is not a valid SIP contact (missing sip:) trying to use anyway [Jul 19 10:46:41] WARNING[2466]: chan_sip.c:8097 set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : '72.16.1.20' [Jul 19 10:46:41] WARNING[2466]: chan_sip.c:5839 set_destination: Can't find address for host '72.16.1.20' [Jul 19 10:46:41] WARNING[2466]: chan_sip.c:5839 set_destination: Can't find address for host '72.16.1.20' Might want to post a sip debug of one of the sessions from the Mitel phone. Thanks Matt I was also able to test this with Mitel's firmware version 7.0.0.8 with the same results. Mitel phone still acts like it's on a call, Asterisk does not nor does the originating phone. PBX*CLI sip set debug peer 517 SIP Debugging Enabled for IP: 172.16.1.174:5060 Audio is at 172.16.1.20 port 15594 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 172.16.1.174:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 172.16.1.20:5060;branch=z9hG4bK38581b5a;rport From: 512 sip:[EMAIL PROTECTED];tag=as7ec9e8af To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sat, 19 Jul 2008 17:20:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 236 v=0 o=root 2247 2247 IN IP4 172.16.1.20 s=session c=IN IP4 172.16.1.20 t=0 0 m=audio 15594 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- PBX*CLI --- SIP read from 172.16.1.174:5060 --- SIP/2.0 100 Trying Via:SIP/2.0/UDP 172.16.1.20:5060;rport=5060;received=172.16.1.20;branch=z9hG4bK38581b5a From:512 sip:[EMAIL PROTECTED];tag=as7ec9e8af To:sip:[EMAIL PROTECTED];tag=4881ea36-2ca-6747d965 CSeq:102 INVITE User-Agent:Mitel-5212-SIP-Phone 07.00.00.08 08000F32A6C1 Call-ID:[EMAIL PROTECTED] Content-Length:0 - --- (8 headers 0 lines) --- PBX*CLI --- SIP read from 172.16.1.174:5060 --- SIP/2.0 180 Ringing Via:SIP/2.0/UDP 172.16.1.20:5060;rport=5060;received=172.16.1.20;branch=z9hG4bK38581b5a From:512 sip:[EMAIL PROTECTED];tag=as7ec9e8af To:sip:[EMAIL PROTECTED];tag=4881ea36-2ca-6747d965 CSeq:102 INVITE User-Agent:Mitel-5212-SIP-Phone 07.00.00.08 08000F32A6C1 Call-ID:[EMAIL PROTECTED] Allow-Events:talk,hold,conference Content-Length:0 - --- (9 headers 0 lines) --- PBX*CLI --- SIP read from 172.16.1.174:5060 --- SIP/2.0 200 OK Via:SIP/2.0/UDP 172.16.1.20:5060;rport=5060;received=172.16.1.20;branch=z9hG4bK38581b5a From:512 sip:[EMAIL PROTECTED];tag=as7ec9e8af To:sip:[EMAIL PROTECTED];tag=4881ea36-2ca-6747d965 CSeq:102 INVITE User-Agent:Mitel-5212-SIP-Phone 07.00.00.08 08000F32A6C1 Call-ID:[EMAIL PROTECTED] Contact:p:[EMAIL PROTECTED];tag=4881ea36-2ca-6747d965 CSeq:102 INVITE User sip:[EMAIL PROTECTED] Allow-Events:talk,hold,conference Allow:INVITE,ACK,CANCEL,BYE,OPTIONS,REFER,NOTIFY,PRACK,UPDATE Supported:timer,100rel,replaces Content-Type:application/sdp Content-Length:182 v=0 o=517 1216473942 1216473941 IN IP4 172.16.1.174 s=SIP Call c=IN IP4 172.16.1.174 t=0 0 m=audio 20012 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 - --- (15 headers 8 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 172.16.1.174:20012 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0x4 (ulaw), 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 172.16.1.174:20012 [Jul 19 13:20:56] NOTICE[2466]: chan_sip.c:8068 set_address_from_contact: 'p:[EMAIL PROTECTED];tag=4881ea36-2ca-6747d965' is not a valid SIP contact (missing sip:) trying to use anyway [Jul 19 13:20:56] WARNING[2466]: chan_sip.c:8097 set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : '172.16.1.174' list_route: hop: p:[EMAIL PROTECTED];tag=4881ea36-2ca-6747d965 set_destination: Parsing p:[EMAIL
Re: [asterisk-users] OT Astricon/Digium Beach Ball Mailing
randulo wrote: On Sat, Jul 19, 2008 at 7:09 PM, Alex Balashov [EMAIL PROTECTED] wrote: I've asked a number of others I know in real life who got the beach balls and all are reported as being fully functional. So this is not a case for the bug tracker? Perhaps a bounty... I've already submitted plastic patches to Beach Ball-rc5-pl5-beta trunk. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Not a valid SIP contact - Asterisk 1.4.21.1 Mitel SIP phones
Maybe stupid solution but, when Mitel phone i called, why dont you pickup put the person on hold, call Mitel phone, and connect them, what i want to say, add some delay. 2008/7/19 Mark Wiater [EMAIL PROTECTED]: Matt Watson wrote: On July 19, 2008 11:22:08 am Mark Wiater wrote: Hi, I have a client using Mitel 5212 phones in SIP mode with a 1.4.21.1 Asterisk server (and a couple of previous 1.4 versions). They're mostly happy with the combination except for this one issue. For incoming calls only, either originating from other local SIP phones or from a PRI, calls won't get bridged (remote party get's hung up) if the call is answer too quickly on the Mitel. Or so it seems. The receiving Mitel phone thinks the call is in session though. Asterisk is reporting errors like: [Jul 19 10:46:41] NOTICE[2466]: chan_sip.c:8068 set_address_from_contact: '72.16.1.20;tag=as7b9f4bfb' is not a valid SIP contact (missing sip:) trying to use anyway [Jul 19 10:46:41] WARNING[2466]: chan_sip.c:8097 set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : '72.16.1.20' [Jul 19 10:46:41] WARNING[2466]: chan_sip.c:5839 set_destination: Can't find address for host '72.16.1.20' [Jul 19 10:46:41] WARNING[2466]: chan_sip.c:5839 set_destination: Can't find address for host '72.16.1.20' Might want to post a sip debug of one of the sessions from the Mitel phone. Thanks Matt I was also able to test this with Mitel's firmware version 7.0.0.8 with the same results. Mitel phone still acts like it's on a call, Asterisk does not nor does the originating phone. PBX*CLI sip set debug peer 517 SIP Debugging Enabled for IP: 172.16.1.174:5060 Audio is at 172.16.1.20 port 15594 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 172.16.1.174:5060: INVITE sip:[EMAIL PROTECTED] [EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 172.16.1.20:5060;branch=z9hG4bK38581b5a;rport From: 512 sip:[EMAIL PROTECTED] [EMAIL PROTECTED];tag=as7ec9e8af To: sip:[EMAIL PROTECTED] [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] [EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sat, 19 Jul 2008 17:20:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 236 v=0 o=root 2247 2247 IN IP4 172.16.1.20 s=session c=IN IP4 172.16.1.20 t=0 0 m=audio 15594 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- PBX*CLI --- SIP read from 172.16.1.174:5060 --- SIP/2.0 100 Trying Via:SIP/2.0/UDP 172.16.1.20:5060;rport=5060;received=172.16.1.20;branch=z9hG4bK38581b5a From:512 sip:[EMAIL PROTECTED] [EMAIL PROTECTED];tag=as7ec9e8af To:sip:[EMAIL PROTECTED] [EMAIL PROTECTED] ;tag=4881ea36-2ca-6747d965 CSeq:102 INVITE User-Agent:Mitel-5212-SIP-Phone 07.00.00.08 08000F32A6C1 Call-ID:[EMAIL PROTECTED][EMAIL PROTECTED] Content-Length:0 - --- (8 headers 0 lines) --- PBX*CLI --- SIP read from 172.16.1.174:5060 --- SIP/2.0 180 Ringing Via:SIP/2.0/UDP 172.16.1.20:5060;rport=5060;received=172.16.1.20;branch=z9hG4bK38581b5a From:512 sip:[EMAIL PROTECTED] [EMAIL PROTECTED];tag=as7ec9e8af To:sip:[EMAIL PROTECTED] [EMAIL PROTECTED] ;tag=4881ea36-2ca-6747d965 CSeq:102 INVITE User-Agent:Mitel-5212-SIP-Phone 07.00.00.08 08000F32A6C1 Call-ID:[EMAIL PROTECTED][EMAIL PROTECTED] Allow-Events:talk,hold,conference Content-Length:0 - --- (9 headers 0 lines) --- PBX*CLI --- SIP read from 172.16.1.174:5060 --- SIP/2.0 200 OK Via:SIP/2.0/UDP 172.16.1.20:5060;rport=5060;received=172.16.1.20;branch=z9hG4bK38581b5a From:512 sip:[EMAIL PROTECTED] [EMAIL PROTECTED];tag=as7ec9e8af To:sip:[EMAIL PROTECTED] [EMAIL PROTECTED] ;tag=4881ea36-2ca-6747d965 CSeq:102 INVITE User-Agent:Mitel-5212-SIP-Phone 07.00.00.08 08000F32A6C1 Call-ID:[EMAIL PROTECTED][EMAIL PROTECTED] Contact:p:[EMAIL PROTECTED] [EMAIL PROTECTED] ;tag=4881ea36-2ca-6747d965 CSeq:102 INVITE User sip:[EMAIL PROTECTED] [EMAIL PROTECTED] Allow-Events:talk,hold,conference Allow:INVITE,ACK,CANCEL,BYE,OPTIONS,REFER,NOTIFY,PRACK,UPDATE Supported:timer,100rel,replaces Content-Type:application/sdp Content-Length:182 v=0 o=517 1216473942 1216473941 IN IP4 172.16.1.174 s=SIP Call c=IN IP4 172.16.1.174 t=0 0 m=audio 20012 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 - --- (15 headers 8 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 172.16.1.174:20012 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities
Re: [asterisk-users] Changinf Polycom-501 config server from remote?
On Saturday 19 July 2008 12:11:46 Michael Graves wrote: There are many parameters that cannot be defined via the web GUI. These must be defined in the config files. Uptake of the configs is only done at boot time. I recall that the www.voip-info.org wiki has a Perl script for remote forcing a reboot of Polycom phones. The command is actually integrated into the CLI: CLI sip notify polycom-check-cfg name Multiple commands like this are possible, and they're all defined in /etc/asterisk/sip_notify.conf -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Echo Issue
This is almost standard with voip calls. The echo-cancellation has to train up to the call parameters. Some hardware is better with it than others and you can try tweaking the value for the echo canceler up and down. What type hardware are you using - both phone and server? On Sat, Jul 19, 2008 at 11:41 AM, Joseph L. Casale [EMAIL PROTECTED] wrote: I am being told by the users on a purely sip based setup that when an inbound sip call is first answered, they here an echo on their greeting and then the conversation stabilizes and it works well. Any ideas where to look to start curing this? Thanks! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 -- Chad Whitten Metro Network Solutions (601) 366-6630 Phone (601) 366-6066 Fax (601) 842-6804 Cellular [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Not a valid SIP contact - Asterisk 1.4.21.1 Mitel SIP phones
2008/7/19 Mark Wiater [EMAIL PROTECTED]: Your problem is the Contact header coming back in the Mitel's Ok response. Contact:p:[EMAIL PROTECTED];tag=4881ea36-2ca-6747d965 Asterisk is not going to be able to parse that! It should be: Contact: sip:[EMAIL PROTECTED];tag=4881ea36-2ca-6747d965 I'd get your Mitel vendor to log a fault about that. It's a big problem as the Contact field is critical for further in-dialogue SIP requests such as transfer related requests and BYEs. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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
[asterisk-users] New Bridge Command/Event in 1.6?
I just downloaded Asterisk 1.6 beta 9 because I had read that there was a new bridge command. After looking through the doc/* documentation, I see no mention of a bridge application or AMI command. Does it exist? I am trying to take a bridged call, and redirect each to another destination, which I can do with the redirect() AMI command. After doing some dial plan processing, I would like to bridge them back together. How can I do this? The redirect command takes a channel and an extension as an argument, not another channel. Doug. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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
[asterisk-users] Stop vm-intro being played
Hi There, Is there a way just to have the custom voice message play, and not have asterisk play: vm-intro after that? Thanks Simon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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
[asterisk-users] Question about stopping Asterisk
Hi all, I've installed Asterisk 1.6 on my Ubuntu Hardy system and I also used the make config command at the end of the installation so that Asterisk loads at boot. However, I want to disable this now. What is the best way of doing this? Many thanks for any help, Christian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Question about stopping Asterisk
Christian wrote: Hi all, I've installed Asterisk 1.6 on my Ubuntu Hardy system and I also used the make config command at the end of the installation so that Asterisk loads at boot. However, I want to disable this now. What is the best way of doing this? Many thanks for any help, Christian This is really an Ubuntu question, but: cd /etc/init.d update-rc.d asterisk remove -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Beginner Questions part II
Hi John, *for the first part: you can create 3 contexts: internal,external and main in your internal context you put your internal extension in the external context you send the send the XXX-XXX- to the providers trunk and in the main context you just include the internal context (first) then the external context. ** second part: check here: http://www.jackenhack.com/blog/archives/2005/09/26/adding-blacklist-to-an-asteriskhome-pbx-voip-server/ *BR,* John Koenig a écrit : I should start with a thank you to the list for helping me getting up and running with Asterisk about a week ago. I have been happily fiddling with Asterisk since then :). I am working on adding a couple features to my dialplan. My setup involves my asterisk box connecting to another third party sip provider. I configured the extra trunk and there are no issues passing calls through their systems. As it stands right now, I setup a calling rule that matches the pattern 9-XXX-XXX-, stripes off the 9, and then passes the call through to the third party. I would like to just dial XXX-XXX- without having to dial the extra 9. Is there a way that I can configure asterisk so that I check to see if the extension exists on my box first, if it does then pickup and if it doesn't then forward the call onto the third party? If so, how? The other feature I am looking to add is *67 caller id blocking. Am I right in thinking that I first would configure an incoming call rule that matches to *67-whatever and then pass unknown as the caller id from there? Any help is greatly appreciated. Even if it is just pointing me in the right direction in regards to reading material. Thanks, John ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] asterisk-users Digest, Vol 48, Issue 57
Hola, Estoy de vacaciones hasta el 1 de Agosto. Para dar soporte sobre la centralita de telefonia: [EMAIL PROTECTED] Perdonen las molestias. Ruth Llaneza Lapausa - Tecnico de VoIP. [EMAIL PROTECTED] Tlf: 902 199 384 Mildmac SA � www.mildmac.es � [EMAIL PROTECTED] C/ Hnos. Garc�a Noblejas 41, 6� planta. 28037 - Madrid Tlf: +34 91 501 33 02 Fax: +34 91 501 57 45 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Echo Issue
This is almost standard with voip calls. The echo-cancellation has to train up to the call parameters. Some hardware is better with it than others and you can try tweaking the value for the echo canceler up and down. Hmm. This has not been my experience. I have rarely seen echo on pure SIP calls, but in all cases that I have, I've found that it is a regular acoustic echo caused by unusual gain settings on at least one end of the call. On Sat, Jul 19, 2008 at 11:41 AM, Joseph L. Casale [EMAIL PROTECTED] wrote: I am being told by the users on a purely sip based setup that when an inbound sip call is first answered, they here an echo on their greeting and then the conversation stabilizes and it works well. Any ideas where to look to start curing this? Thanks! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 -- Chad Whitten Metro Network Solutions (601) 366-6630 Phone (601) 366-6066 Fax (601) 842-6804 Cellular [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Beep on transfer
Hi John - I have a request that I have not been able to figure out as yet. I need to be able to play a beep when a call is transfered via attended transfer. This is exactly what is in the bug tracker at: http://bugs.digium.com/view.php?id=3819 Has any one found a way, elegant ot otherwise, to make something such as this work? Thanks in advance for any help. Here's an incredibly inelegant way: When an incoming call hits an extension, set a channel variable to a particular value. Put a check in your extension logic to see if that channel variable is set (put this before you set the channel variable). If the variable is set, play a beep. If it's not, don't play a beep. Extremely hackish, but it would fulfill the request. Yes, this would play a beep if a call was just blind transferred. It would also beep if the call was parked (and possibly picked up by the same person), but you could also hack some more to avoid this. - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Stop vm-intro being played
Got it!... we are using Elastix and i just had to set s in the VM_OPTS in the extensions_additional.conf file. On Sun, Jul 20, 2008 at 1:41 PM, Simon [EMAIL PROTECTED] wrote: Hi There, Is there a way just to have the custom voice message play, and not have asterisk play: vm-intro after that? Thanks Simon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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
[asterisk-users] conference bridge
Hi, How can i setup conference when i have 2 asterisk servers? my setup is 2 asterisk servers using realtime, i'm simply using DNS SRV just for redundancy (not really high availability). i have a web interface, wherein i can create extension, conference etc. adding extension is ok, even if ext1 is registered on Asterisk 1 and ext2 is registered on asterisk 2 they will still be able to call each other, but on the conference, e.g. when ext1 dials conference no. 1000 and ext 2 dials conf 1000 also, they will be connected to two different conference room. my meetme is also setup on realtime. how can i set it up in such a way ext on registered on different asterisk server can connect to the same conference room. Regrdas, Nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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
