Re: [asterisk-users] How to see STDERR message?
Thank you for you reply? is that mean STDERR couldn't show under Asterisk CLI mode? it's only saved to some file? 2010/1/7 Steve Edwards asterisk@sedwards.com: On Thu, 7 Jan 2010, Zhang Shukun wrote: i use agi to send message back to Asterisk by STDERR, but why i could't see the message in asterisk CLI? Output to STDERR does nothing for me either. I prefer to use syslog() to log the messages via syslogd. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Sucan ___ -- 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] error compile dahdi with latest kernels.
hello, all of users: there are header files missed when you compile dahdi with kernel-2.6.29 or 2.6.33. i believe that few files are affected: wctdm.c dahdi-base.c wcb4xxp/base.c, opvxa1200.c... the errors look like these: from /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c:61: /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/include/dahdi/dahdi_config.h:27:28: error: linux/autoconf.h: No such file or directory /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c: In function '__qevent': /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c:839: error: 'TASK_INTERRUPTIBLE' undeclared (first use in this function) /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c:839: error: (Each undeclared identifier is reported only once /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c:839: error: for each function it appears in.) /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c: In function 'schluffen': /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c:867: error: dereferencing pointer to incomplete type /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c:867: error: 'TASK_INTERRUPTIBLE' undeclared (first use in this function) /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c:869: error: implicit declaration of function 'signal_pending' /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c:870: error: implicit declaration of function 'schedule' /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c:872: error: dereferencing pointer to incomplete type /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c:872: error: 'TASK_RUNNING' undeclared (first use in this function) /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c: In function 'dahdi_timer_ioctl': /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c:3418: error: 'TASK_INTERRUPTIBLE' undeclared (first use in this function) /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c: In function 'dahdi_chanandpseudo_ioctl': /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c:4419: error: 'TASK_INTERRUPTIBLE' undeclared (first use in this function) /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c: In function '__dahdi_getbuf_chunk': /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c:6075: error: 'TASK_INTERRUPTIBLE' undeclared (first use in this function) /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c: In function '__rbs_otimer_expire': /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c:6263: error: 'TASK_INTERRUPTIBLE' undeclared (first use in this function) /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c: In function '__putbuf_chunk': /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c:7203: error: 'TASK_INTERRUPTIBLE' undeclared (first use in this function) /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c: In function 'dahdi_hdlc_finish': == after digging the code, i changed the files and add some linux headers. #include linux/kernel.h #include linux/errno.h +#include linux/sched.h #include linux/module.h #include linux/proc_fs.h = and add this: #ifdef __KERNEL__ #include linux/version.h #if LINUX_VERSION_CODE KERNEL_VERSION(2,6,18) #include linux/config.h #else +#include generated/autoconf.h -#include linux/autoconf.h #endif #endif = Regards! zhulizhong ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to see STDERR message?
Hello, STDERR goes to the original Asterisk process only, not any asterisk -r connections that you may use. If you launch Asterisk in a screen like we do, then you can see it and log it in context with when the output is happening. We find it very useful to do it this way. MATT--- On 1/7/10, Zhang Shukun bit...@gmail.com wrote: Thank you for you reply? is that mean STDERR couldn't show under Asterisk CLI mode? it's only saved to some file? 2010/1/7 Steve Edwards asterisk@sedwards.com: On Thu, 7 Jan 2010, Zhang Shukun wrote: i use agi to send message back to Asterisk by STDERR, but why i could't see the message in asterisk CLI? Output to STDERR does nothing for me either. I prefer to use syslog() to log the messages via syslogd. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Sucan ___ -- 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] error compile dahdi with latest kernels.
On Thu, Jan 07, 2010 at 04:19:21PM +0800, james.zhu wrote: hello, all of users: there are header files missed when you compile dahdi with kernel-2.6.29 or 2.6.33. i believe that few files are affected: wctdm.c dahdi-base.c wcb4xxp/base.c, opvxa1200.c... the errors look like these: from /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c:61: /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/include/dahdi/dahdi_config.h:27:28: error: linux/autoconf.h: No such file or directory http://svnview.digium.com/svn/dahdi?view=revisionrevision=7732 -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/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] DTMF detection on dahdi with b4xxp (again, some more details)
Hi, Am Dienstag, den 05.01.2010, 15:38 +0100 schrieb Christian Theune: Hi, I tried again getting DTMF detection on my ISDN devices with dahdi going again. I used the channel debug to see whether asterisk sees the frames and detects them as DTMF. Interestingly here's what works: 1. GSM phone - chan_dahdi g1 - asterisk - can_sip - SIP phone Both the GSM phone and the SIP phone can issue DTMF that will be detected as features (transfer) 2. GSM phone - chan_dahdi g1 - asterisk - chan_dahdi g4 - ISDN phone The GSM phone can issue DTMF that will be detected. The ISDN phone won't. (That's my issue.) I don't see any messages of asterisk recognizing the DTMF frames when pressing the keys. I do hear the DMTF sound on both phones. 3. ISDN phone - chan_dahdi g4 - asterisk - chan_dahdi g1 - GSM phone The ISDN phone can issue DTMF that will be recognized and so does the GSM phone. So. When the ISDN phone is receiving a call on g4 its DTMF sounds won't be recognized. OTOH when the GSM phone on g1 is being called it's sounds are recognized. I *think* there are two possibilities to transfer DTMF on ISDN: - as audio on B-Channel - as Key-Press events (Info-Elements) on D-Channel DTMF on GSM can not be signalled as audio (because of codec with high compression). I guess in case GSM = asterisk via chan_dahdi g1 in Your example, the DTMF is signalled as Info-Elements on D-Channel. I guess in the cases where Your DTMF is not working, audio path is used. In this case DTMF detection is done by DSP-Software. Look for the relaxdtmf statement (in case of zaptel this worked for me in a simmilar scenario). HTH, Karsten ___ -- 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] caller getting cut off intermittently
We're now getting this problem on outgoing calls. I've forced the port to 100FD but still no joy. Anyone any ideas how to debug this- have added verbose to logger.conf Thanks for any help John 2010/1/4 John Taylor j...@vetsurgeon.org.uk: I have recently moved our asterisk server from our LAN to a Debian Lenny server (Asterisk 1.4.21.2~dfsg-3 ) with a public IP outside our network. Our phones are behind a natted firewall. An ITSP provides a PSTN to SIP termination for incoming calls Public ITSP --Asterisk server--Natted firewall--extension (192.168.1.x) Everything works fine (incoming/outgoing audio etc.) except occasionally an incoming caller is cut off whilst the called extension stays in the call and can hear a DTMF tone (multimon recognises it as tone D). The asterisk log file shows the call stays active despite the incoming caller being cut off. This has happened to all our extensions at some point (a combination of Snoms and Funkwerks). It happens fairly infrequently, and can happen at any point during a call. The public Lenny server's asterisk config is exactly the same as our LAN Ubuntu asterisk server where we never had this problem. The only difference is that the ITSP trunk is now ulaw rather than ilbc. Can anyone help? Relevant files below (trunk and extension codecs are both ulaw) John example extension in sip.conf: [203] type=friend username=203 secret=xx host=dynamic dtmfmode=inband call-limit=2 qualify=yes nat=yes /var/log/asterisk/messages: [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [301xx...@fromvoipfone:1] Set(SIP/301x-09f74a00, oh=0) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [301xx...@fromvoipfone:2] NoOp(SIP/301x-09f74a00, 01295259352) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [301xx...@fromvoipfone:3] GotoIf(SIP/301x-09f74a00, 0?bankhols|200|1) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [301xx...@fromvoipfone:4] GotoIfTime(SIP/301x-09f74a00, 08:30-18:00|mon-fri|*|*?day|100|1) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Goto (day,100,1) [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [...@day:1] AGI(SIP/301x-09f74a00, /home/john/phpagi/lookup) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Launched AGI Script /home/john/phpagi/lookup [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- AGI Script /home/john/phpagi/lookup completed, returning 0 [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [...@day:2] Set(SIP/301x-09f74a00, CALLERID(name)=) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [...@day:3] Macro(SIP/301x-09f74a00, monitor|01327xx|in) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [...@macro-monitor:1] Set(SIP/301x-09f74a00, CALLFILENAME=/home/john/asterisk/asterisk_recordings/in-20100104_095856-01295259352) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [...@macro-monitor:2] Monitor(SIP/301x-09f74a00, wav|/home/john/asterisk/asterisk_recordings/in-20100104_095856-01295259352|m) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [...@day:4] Dial(SIP/301x-09f74a00, SIP/203SIP/204SIP/206SIP/207SIP/220SIP/221|20|t) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Called 203 [Jan 4 09:58:56] WARNING[10712] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Called 206 [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Called 207 [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Called 220 [Jan 4 09:58:56] WARNING[10712] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- SIP/220-09fe7748 is ringing [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- SIP/206-0a005eb8 is ringing [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- SIP/207-09fe2c98 is ringing [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- SIP/203-0a001138 is ringing [Jan 4 09:58:57] VERBOSE[10712] logger.c: -- SIP/220-09fe7748 is ringing [Jan 4 09:58:57] VERBOSE[10712] logger.c: -- SIP/203-0a001138 is ringing [Jan 4 09:58:58] VERBOSE[10712] logger.c: -- SIP/220-09fe7748 is ringing [Jan 4 09:58:58] VERBOSE[10712] logger.c: -- SIP/203-0a001138 is ringing [Jan 4 09:58:59] VERBOSE[10712] logger.c: -- SIP/203-0a001138 answered SIP/301x-09f74a00 ___ -- 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] No reply to SIP OPTIONS - sip peers becoming randomly UNREACHABLE
Hi, I have occasionally experienced the same problem too, and I suspect it was caused by some spikes in network traffic (e.g. for an intensive file transfer) that delayed too much SIP OPTION response, so that Asterisk marked these devices as UNREACHABLE; I was able to use the devices too: in fact, the only drawback is that other devices are not able to call the UNREACHABLE devices using Asterisk. The only solution I found was to disable 'qualify' field in SIP account, in order to put these devices in unmonitored state. Maybe it's not your problem, but you can monitor the network with a sniffer (e.g. ethereal), in conjunction with SIP debug in Asterisk (sip set debug) in order to check the correct arrival of OPTION response. Noevertheless, I'm wondering if there is another cause to this issue that is not depending on network, but on Asterisk itself, so let me know. HTH, cheers Alberto. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk Sent: lunedì 4 gennaio 2010 22.13 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] No reply to SIP OPTIONS - sip peers becoming randomly UNREACHABLE Hi guys, Am having a strange SIP problem in my call centre. The call centre has about 70 SIP agents (some of the are using SIP hard phones, other SIP softphones), and occasionally most of the SIP peers (hardphones and softphones) become UNREACHABLE and then after few second again REACHABLE. Some hardphones and softphones work perfectly normal during that period (even normally responding to OPTIONS message), but most of them get UNREACHABLE. I don't have NAT - phones and Asterisk are in the same subnet, so nothing complicated really (regarding network configuration). I'm currently suspecting my network to be the problem, but I would just like to confirm with you guys, if you have any similar experiences, what could be causing this? Please, see bellow one of the sample SIP traces. Regards, Alex Jan 1 11:17:42 VERBOSE[6046] logger.c: Reliably Transmitting (no NAT) to 165.11.1.41:5060: OPTIONS sip:testpho...@165.11.1.41 SIP/2.0 Via: SIP/2.0/UDP 165.11.1.50:5060;branch=a4bG4bK4b7hf375;rport From: asterisk sip:aster...@165.11.1.50;tag=as02e1afaa To: sip:testpho...@165.11.1.41 Contact: sip:aster...@165.11.1.50 Call-ID: 3fa169320586bad01cd93bd87adf1...@165.11.1.50 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 01 Jan 2010 11:17:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 Jan 1 11:17:45 VERBOSE[6046] logger.c: Retransmitting #1 (no NAT) to 165.11.1.41:5060: OPTIONS sip:testpho...@165.11.1.41 SIP/2.0 Via: SIP/2.0/UDP 165.11.1.50:5060;branch=a4bG4bK4b7hf375;rport From: asterisk sip:aster...@165.11.1.50;tag=as02e1afaa To: sip:testpho...@165.11.1.41 Contact: sip:aster...@165.11.1.50 Call-ID: 3fa169320586bad01cd93bd87adf1...@165.11.1.50 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 01 Jan 2010 11:17:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 Jan 1 11:17:46 NOTICE[6046] chan_sip.c: Peer 'TestPhone1' is now UNREACHABLE! Last qualify: 14 Jan 1 11:17:56 VERBOSE[6046] logger.c: Reliably Transmitting (no NAT) to 165.11.1.41:5060: OPTIONS sip:testpho...@165.11.1.41 SIP/2.0 Via: SIP/2.0/UDP 165.11.1.50:5060;branch=z2h16b637dKh2fd;rport From: asterisk sip:aster...@165.11.1.50;tag=as796f6356 To: sip:testpho...@165.11.1.41 Contact: sip:aster...@165.11.1.50 Call-ID: 3367c4dc6cbdd57d67b0c5b53d549...@165.11.1.50 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 01 Jan 2010 11:17:56 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 Jan 1 11:17:56 VERBOSE[6046] logger.c: -- SIP read from 165.11.1.41:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 165.11.1.50:5060;branch=z2h16b637dKh2fd;rport From: asterisk sip:aster...@165.11.1.50;tag=as796f6356 To: sip:testpho...@165.11.1.41;tag=5A4BF5F8-460290A9 CSeq: 102 OPTIONS Call-ID: 3367c4dc6cbdd57d67b0c5b53d549...@165.11.1.50 Contact: sip:testpho...@165.11.1.41 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.2.0.0047 Content-Length: 0 Jan 1 11:17:56 VERBOSE[6046] logger.c: --- (10 headers 0 lines) --- Jan 1 11:17:56 NOTICE[6046] chan_sip.c: Peer 'TestPhone1' is now REACHABLE! (16ms / 1ms) ___ -- 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:
Re: [asterisk-users] Really Silly Question From Total Newbie
Hello there! If your box has a live Internet connection, then all you need is a sip provider. Back to when I lived in the UK, there was this voipuser.org which gave me a fixed british number for free, and some outbound call minutes too. I'm sure that if you search around for SIP Providers, you may be able to find some free stuff. I believe that Outbound calls cost money, not incoming calls. I'm not totally sure tho. Anyway, you should find a provider and try to register with them, - Regards, Tiago Lourenço Geada 2010/1/5 UIT DEVELOPMENT uit...@gmail.com Jamie - I will check that out! Thanks! It is just for testing and yes, the Asterisk box is connected to the Internet. Cool. -M On Tue, Jan 5, 2010 at 4:39 PM, Jamie A. Stapleton jstaple...@computer-business.com wrote: Could use the free http://www.sipgate.com/one for some testing (assuming that Asterisk is connected to the Internet) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of UIT DEVELOPMENT Sent: Tuesday, January 05, 2010 2:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Really Silly Question From Total Newbie Hello All - I've been poking around the past few weeks, trying to familiarize myself with all of this. I am new to Linux, VoIP and Asterisk -- to be complete. This is my first exposure to all of these technologies. I installed AsteriskNow on my old dual Pentium 833mhz Dell PowerEdge 2400 and the install went well. I can log in and poke around in Linux and I even configured the box to be recognized on my windows network. However, is there a GUI that I can access to help me set things up? I've gotten so far as what looks to me like DOS windows that I can change various things in the OS... I do not have any other hardware installed. No cards and no VoIP phones. I havent got to the point where I can make a test call or anything like that. I dont know how to tell if Asterisk is up and running and how I can tweak it, etc. I've been reading a lot of different things, and have become a bit confused. I think that in time it will come to me but I needed to stop and ask because I need to know if I am on the wrong path for what I'd like to do someday My main question is: CAN I make call from that box to my cell phone using a soft-phone? If so, how can I do that? Also, can I use my cell phone to call into that box? I dont know if I have to get a phone number, or do I NEED a phone number? At the moment, I do not have any dollars to throw at this project. Its purely for learning, proof of concept sort of thing for myself on my spare time in the evenings. I would simply like to be able to call out and be able to call into that box. Later on down the road maybe I will get into setting up an IVR using a database so I can call into that system from wherever and get information read back to me. But, first things first I'd like to know if I am heading down the wrong path here. Sorry for what might seem as really silly questions, but I am not sure how to proceed. Thanks in advance for any insight that you folks can provide! Mike ___ -- 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 ___ -- 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] iaxmodem to ReceiveFAX crashes Asterisk equipped with B410P
2010/1/7 David Backeberg dbackeb...@gmail.com On Wed, Jan 6, 2010 at 6:23 PM, Olivier oza-4...@myamail.com wrote: The second time I'm dialing an internal extension attached to the same ReceiveFAX application : 2. sendfax/hylafax/iaxmodem asterisk spandsp In the 2nd case, I've got 3 craches out of 3 attempts (with a rough estimee, the crash occurs 2 or 4s after ReceiveFAX's start). Before wasting anybody's time and effort within Asterisk support team, I would like to double check here if the case that crashes Asterisk is within specifications of involved apps. In other words, can you normally use Hylafax to send faxes to inner extensions or do you hace to stick to PSTN numbers ? I've never successfully done what you're trying to do, so I came to the conclusion that it was not supported. My reasoning is exactly the same as yours. It would be great if a developer could drop in and tell if this feature is supported or not. When I wanted to test faxing, I ended up using two systems, sometimes with just LAN in-between and sometimes with PSTN in-between. ___ -- 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] iaxmodem to ReceiveFAX crashes Asterisk equipped with B410P
Olivier schrieb: 2010/1/7 David Backeberg dbackeb...@gmail.com mailto:dbackeb...@gmail.com On Wed, Jan 6, 2010 at 6:23 PM, Olivier oza-4...@myamail.com mailto:oza-4...@myamail.com wrote: The second time I'm dialing an internal extension attached to the same ReceiveFAX application : 2. sendfax/hylafax/iaxmodem asterisk spandsp In the 2nd case, I've got 3 craches out of 3 attempts (with a rough estimee, the crash occurs 2 or 4s after ReceiveFAX's start). Before wasting anybody's time and effort within Asterisk support team, I would like to double check here if the case that crashes Asterisk is within specifications of involved apps. In other words, can you normally use Hylafax to send faxes to inner extensions or do you hace to stick to PSTN numbers ? I've never successfully done what you're trying to do, so I came to the conclusion that it was not supported. My reasoning is exactly the same as yours. It would be great if a developer could drop in and tell if this feature is supported or not. What asterisk version and spandsp version do you use ? IMO a crash can never be an answer if this kind of setup is supported or not. regards Hans ___ -- 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] Explain what asterisk.conf's internal timing option is
Hello, I've read in Mantis that asterisk.conf's internal timing option could positively impact Asterisk behaviour during faxing ( http://issues.asterisk.org/view.php?id=16374). Before using it, I would be very pleased to read a line or two about its use. I've read http://www.russellbryant.net/blog/2008/06/16/asterisk-16-now-with-a-new-timing-api/but I still a couple of questions. When you have a 1.6.1 server with a PSTN trunk, is this option of any use, as in my opinion, timing is then provided by PSTN ? Regards ___ -- 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] queue and linear strategy
Hello, I've upgraded asterisk to 1.6.0.20 version and found , if I want change queue strategy to linear, I must restart Asterisk: [Jan 7 08:16:10] WARNING[9578]: app_queue.c:1304 queue_set_param: Changing to the linear strategy currently requires asterisk to be restarted. [Jan 7 08:16:10] WARNING[9578]: app_queue.c:1304 queue_set_param: Changing to the linear strategy currently requires asterisk to be restarted. Maybe somebody can explain, why I need restart asterisk? Thanks -- Pagarbiai / Best Regards, Giedrius ___ -- 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] HDLC Receiver overrun
Can anyone shed any light on this error? Will Jan 7 11:03:34 asterisk pppd[9168]: Plugin zaptel.so loaded. Jan 7 11:03:34 asterisk pppd[9168]: Zaptel Plugin Initialized Jan 7 11:03:34 asterisk pppd[9168]: Using zaptel device 'stdin' Jan 7 11:03:34 asterisk pppd[9168]: pppd 2.4.4 started by root, uid 0 Jan 7 11:03:34 asterisk pppd[9168]: Zaptel device is 'stdin' Jan 7 11:03:34 asterisk pppd[9168]: Connected to zaptel device 'WCT1/0/1' (65537) Jan 7 11:03:34 asterisk pppd[9168]: Using interface ppp0 Jan 7 11:03:34 asterisk pppd[9168]: Connect: ppp0 -- stdin Jan 7 11:03:35 asterisk kernel: HDLC Receiver overrun on channel WCT1/0/1 (master=WCT1/0/1) Jan 7 11:03:36 asterisk pppd[9168]: Terminating on signal 15 Jan 7 11:03:42 asterisk pppd[9168]: Connection terminated. Jan 7 11:03:42 asterisk pppd[9168]: Disconnect from zaptel Jan 7 11:03:42 asterisk pppd[9168]: Modem hangup Jan 7 11:03:42 asterisk pppd[9168]: Exit. ___ -- 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] iaxmodem to ReceiveFAX crashes Asterisk equipped with B410P
2010/1/7 Johann Steinwendtner steinwendt...@gmx.net Olivier schrieb: 2010/1/7 David Backeberg dbackeb...@gmail.com mailto:dbackeb...@gmail.com On Wed, Jan 6, 2010 at 6:23 PM, Olivier oza-4...@myamail.com mailto:oza-4...@myamail.com wrote: The second time I'm dialing an internal extension attached to the same ReceiveFAX application : 2. sendfax/hylafax/iaxmodem asterisk spandsp In the 2nd case, I've got 3 craches out of 3 attempts (with a rough estimee, the crash occurs 2 or 4s after ReceiveFAX's start). Before wasting anybody's time and effort within Asterisk support team, I would like to double check here if the case that crashes Asterisk is within specifications of involved apps. In other words, can you normally use Hylafax to send faxes to inner extensions or do you hace to stick to PSTN numbers ? I've never successfully done what you're trying to do, so I came to the conclusion that it was not supported. My reasoning is exactly the same as yours. It would be great if a developer could drop in and tell if this feature is supported or not. What asterisk version and spandsp version do you use ? 1.6.1.11 and 0.0.6pre12 I've seen a couples of rissues in Mantis ( https://issues.asterisk.org/view.php?id=16361 for instance) related to faxing but nothing for iaxmodem/ReceiveFAX interop. IMO a crash can never be an answer if this kind of setup is supported or not. Yes, I agree with that (Asterisk shouldn't crash at all) but I also think that it's not practical to develop a system so robust that it won't crash if the environment or usage is outside its specs. regards Hans ___ -- 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] Really Silly Question From Total Newbie
Hello Tiago, I think that this is the route I will be trying to go as its a proof of concept sort of project. After that - we'll see. Thank you! On Thu, Jan 7, 2010 at 4:43 AM, Tiago Geada tiago.ge...@gmail.com wrote: Hello there! If your box has a live Internet connection, then all you need is a sip provider. Back to when I lived in the UK, there was this voipuser.org which gave me a fixed british number for free, and some outbound call minutes too. I'm sure that if you search around for SIP Providers, you may be able to find some free stuff. I believe that Outbound calls cost money, not incoming calls. I'm not totally sure tho. Anyway, you should find a provider and try to register with them, - Regards, Tiago Lourenço Geada 2010/1/5 UIT DEVELOPMENT uit...@gmail.com Jamie - I will check that out! Thanks! It is just for testing and yes, the Asterisk box is connected to the Internet. Cool. -M On Tue, Jan 5, 2010 at 4:39 PM, Jamie A. Stapleton jstaple...@computer-business.com wrote: Could use the free http://www.sipgate.com/one for some testing (assuming that Asterisk is connected to the Internet) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of UIT DEVELOPMENT Sent: Tuesday, January 05, 2010 2:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Really Silly Question From Total Newbie Hello All - I've been poking around the past few weeks, trying to familiarize myself with all of this. I am new to Linux, VoIP and Asterisk -- to be complete. This is my first exposure to all of these technologies. I installed AsteriskNow on my old dual Pentium 833mhz Dell PowerEdge 2400 and the install went well. I can log in and poke around in Linux and I even configured the box to be recognized on my windows network. However, is there a GUI that I can access to help me set things up? I've gotten so far as what looks to me like DOS windows that I can change various things in the OS... I do not have any other hardware installed. No cards and no VoIP phones. I havent got to the point where I can make a test call or anything like that. I dont know how to tell if Asterisk is up and running and how I can tweak it, etc. I've been reading a lot of different things, and have become a bit confused. I think that in time it will come to me but I needed to stop and ask because I need to know if I am on the wrong path for what I'd like to do someday My main question is: CAN I make call from that box to my cell phone using a soft-phone? If so, how can I do that? Also, can I use my cell phone to call into that box? I dont know if I have to get a phone number, or do I NEED a phone number? At the moment, I do not have any dollars to throw at this project. Its purely for learning, proof of concept sort of thing for myself on my spare time in the evenings. I would simply like to be able to call out and be able to call into that box. Later on down the road maybe I will get into setting up an IVR using a database so I can call into that system from wherever and get information read back to me. But, first things first I'd like to know if I am heading down the wrong path here. Sorry for what might seem as really silly questions, but I am not sure how to proceed. Thanks in advance for any insight that you folks can provide! Mike ___ -- 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 ___ -- 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] iaxmodem to ReceiveFAX crashes Asterisk equipped with B410P
PS: If you compile Asterisk from source after installing spandsp, SendFAX and ReceiveFAX would automatically be included. I opened another thread about that but I doubt that both SendFAX and ReceiveFAX behave exactly the same (no side effect), no matter the installed spandsp version. I would be very happy to be proven to be wrong on this but, the strange thing is I couldn't find anywhere within Asterisk source file, a note mentioning which spandsp's version is supported and which is not. ___ -- 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] No reply to SIP OPTIONS - sip peers becoming randomly UNREACHABLE
7 jan 2010 kl. 10.21 skrev Aggio Alberto: Hi, I have occasionally experienced the same problem too, and I suspect it was caused by some spikes in network traffic (e.g. for an intensive file transfer) that delayed too much SIP OPTION response, so that Asterisk marked these devices as UNREACHABLE; I was able to use the devices too: in fact, the only drawback is that other devices are not able to call the UNREACHABLE devices using Asterisk. The only solution I found was to disable 'qualify' field in SIP account, in order to put these devices in unmonitored state. Maybe it's not your problem, but you can monitor the network with a sniffer (e.g. ethereal), in conjunction with SIP debug in Asterisk (sip set debug) in order to check the correct arrival of OPTION response. Noevertheless, I'm wondering if there is another cause to this issue that is not depending on network, but on Asterisk itself, so let me know. The interesting thing to check here is if you can place a call TO the phone while it's marked as UNREACHABLE. Unreachable means that asterisk has sent an OPTIONs message 7 times without getting a reply at all. If it's LAGGED, we've got a reply, but far too late. If we can't get multiple OPTIONs through, how can we get an INVITE through? /O ___ -- 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] Explain what asterisk.conf's internal timing option is
7 jan 2010 kl. 12.00 skrev Olivier: Hello, I've read in Mantis that asterisk.conf's internal timing option could positively impact Asterisk behaviour during faxing (http://issues.asterisk.org/view.php?id=16374). Before using it, I would be very pleased to read a line or two about its use. I've read http://www.russellbryant.net/blog/2008/06/16/asterisk-16-now-with-a-new-timing-api/ but I still a couple of questions. When you have a 1.6.1 server with a PSTN trunk, is this option of any use, as in my opinion, timing is then provided by PSTN ? If you have any DAHDI/Zaptel driver, then you have timing. The internal timing option in asterisk.conf will affect the RTP flow in calls that use RTP. (XMPP/Jingle, SIP, MGCP, H.323). Since the RTP system is normally clocked on incoming packets, there are issues where you will reach a standstill - no one is sending any packets, because both systems are waiting for a packet to arrive. Internal timing will then take over and force Asterisk to send packets based on the Dahdi/Zaptel timer and not based on the incoming flow. /O ___ -- 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] Zaptel compilation problems: Data Mode!!
Nobody can help me on this?? -- Hi all, I want to compile zaptel in data mode but i got this errors: /usr/src/zaptel-1.4.12.1/kernel/zaptel-base.c: In function âzt_xmitâ: /usr/src/zaptel-1.4.12.1/kernel/zaptel-base.c:1618: error: implicit declaration of function âhdlc_statsâ /usr/src/zaptel-1.4.12.1/kernel/zaptel-base.c:1618: warning: initialization makes pointer from integer without a cast /usr/src/zaptel-1.4.12.1/kernel/zaptel-base.c: In function âzt_ppp_xmitâ: /usr/src/zaptel-1.4.12.1/kernel/zaptel-base.c:1722: warning: comparison of distinct pointer types lacks a cast /usr/src/zaptel-1.4.12.1/kernel/zaptel-base.c:1785: warning: comparison of distinct pointer types lacks a cast /usr/src/zaptel-1.4.12.1/kernel/zaptel-base.c: In function â__putbuf_chunkâ: /usr/src/zaptel-1.4.12.1/kernel/zaptel-base.c:6806: warning: initialization makes pointer from integer without a cast /usr/src/zaptel-1.4.12.1/kernel/zaptel-base.c:6819: warning: initialization makes pointer from integer without a cast /usr/src/zaptel-1.4.12.1/kernel/zaptel-base.c:6912: warning: initialization makes pointer from integer without a cast make[3]: *** [/usr/src/zaptel-1.4.12.1/kernel/zaptel-base.o] Error 1 make[2]: *** [_module_/usr/src/zaptel-1.4.12.1/kernel] Error 2 make[2]: Leaving directory `/usr/src/linux-2.6.27.7' make[1]: *** [modules] Error 2 make[1]: Leaving directory `/usr/src/zaptel-1.4.12.1' make: *** [all] Error 2 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to dial a number make two phone Ring at the same time?
hi, i want to dial a number to let two phone ring at the same time or alternative ring, how should i configure in asterisk? or how to right the Dialplan code? Thanks very much! -- Best regards, Sucan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to dial a number make two phone Ring at the same time?
On Thu, Jan 7, 2010 at 2:38 PM, Zhang Shukun bit...@gmail.com wrote:
hi,
i want to dial a number to let two phone ring at the same time or
alternative ring,
how should i configure in asterisk? or how to right the Dialplan code?
exten = 12345,1,Dial(${PHONE1}${PHONE2})
each phone variable is defined as stated in docs depending on the
protocol, SIP, IAX2, etc
as in
exten = s,1,Dial(SIP/2000)
So PHONE1 would be SIP/2000
See
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.con
/r
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Re: [asterisk-users] iaxmodem to ReceiveFAX crashes Asterisk equipped with B410P
On 01/07/2010 07:21 PM, Olivier wrote: PS: If you compile Asterisk from source after installing spandsp, SendFAX and ReceiveFAX would automatically be included. I opened another thread about that but I doubt that both SendFAX and ReceiveFAX behave exactly the same (no side effect), no matter the installed spandsp version. I would be very happy to be proven to be wrong on this but, the strange thing is I couldn't find anywhere within Asterisk source file, a note mentioning which spandsp's version is supported and which is not. The commonest cause of this kind of problem is having multiple versions of spandsp installed in different directories - e.g /usr and /usr/local. The funky build systems can manage to build against one, but run using the other one. If the aren't binary compatible.. boom. As far as I know, app_fax adapts at build time to the version of spandsp, for any version of spandsp from the last couple of years. You really ought to be running spandsp-0.0.6pre16 for best results. Regards, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dialing OutBound SIP trunk using Dial() command
Hello users,
i am working on directly calling the numbers from the sip provider of my
choice from asterisk using Dial command as follows.
extensions.conf
[dial-out]
exten = _XX,1,NoOp(Dialing out)
exten =
_XX,n,Dial(SIP/1{EXTEN}:password:md5secret:authname:tarnsp...@host:port
, 20,r)
exten = _XX,n,Hangup()
//so i am trying to call the number using voip provider details i have
but i am getting the following error in asterisk CLI
SIP/408XXX:x::XXX:u...@xx
Called 140:x::XXX:u...@xx
-- SIP/xx-0a155070 is circuit-busy
when i try with other service provider i am getting a similar error in
asterisk CLI
SIP/1408X:y::YY:u...@yyy
Got SIP response 500 Nice try back from 64.xx.xx.xx
-- SIP/yyy-0a16ac20 is circuit-busy
my idea is to allow users to enter their own voip providers for outgoing
calls
so that customer can use his own voip provider
i am NOT LOOKING FOR A SOLUTION in /etc/sip.conf entries
like
register = username:passw...@myprovider
[myprovider]
username=
secret=
fromuser=
fromdomain=
host=
any help is appreciated.
Thanks
srinvias
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Re: [asterisk-users] How to dial a number make two phone Ring at the same time?
Thank you!
but how can i determine whether ring at the same time or
alternative ring?
BTW, the uri
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.con
can't open.
Could you paste it again?
2010/1/7 Randy R randulo2...@gmail.com:
On Thu, Jan 7, 2010 at 2:38 PM, Zhang Shukun bit...@gmail.com wrote:
hi,
i want to dial a number to let two phone ring at the same time or
alternative ring,
how should i configure in asterisk? or how to right the Dialplan code?
exten = 12345,1,Dial(${PHONE1}${PHONE2})
each phone variable is defined as stated in docs depending on the
protocol, SIP, IAX2, etc
as in
exten = s,1,Dial(SIP/2000)
So PHONE1 would be SIP/2000
See
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.con
/r
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--
Best regards,
Sucan
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Re: [asterisk-users] Zaptel compilation problems: Data Mode!!
On Thu, Jan 07, 2010 at 07:27:00AM -0600, mos...@infolog.mr wrote: Nobody can help me on this?? -- Hi all, I want to compile zaptel in data mode but i got this errors: /usr/src/zaptel-1.4.12.1/kernel/zaptel-base.c: In function âzt_xmitâ: /usr/src/zaptel-1.4.12.1/kernel/zaptel-base.c:1618: error: implicit declaration of function âhdlc_statsâ /usr/src/zaptel-1.4.12.1/kernel/zaptel-base.c:1618: warning: initialization makes pointer from integer without a cast /usr/src/zaptel-1.4.12.1/kernel/zaptel-base.c: In function âzt_ppp_xmitâ: /usr/src/zaptel-1.4.12.1/kernel/zaptel-base.c:1722: warning: comparison of distinct pointer types lacks a cast /usr/src/zaptel-1.4.12.1/kernel/zaptel-base.c:1785: warning: comparison of distinct pointer types lacks a cast /usr/src/zaptel-1.4.12.1/kernel/zaptel-base.c: In function â__putbuf_chunkâ: /usr/src/zaptel-1.4.12.1/kernel/zaptel-base.c:6806: warning: initialization makes pointer from integer without a cast /usr/src/zaptel-1.4.12.1/kernel/zaptel-base.c:6819: warning: initialization makes pointer from integer without a cast /usr/src/zaptel-1.4.12.1/kernel/zaptel-base.c:6912: warning: initialization makes pointer from integer without a cast make[3]: *** [/usr/src/zaptel-1.4.12.1/kernel/zaptel-base.o] Error 1 make[2]: *** [_module_/usr/src/zaptel-1.4.12.1/kernel] Error 2 make[2]: Leaving directory `/usr/src/linux-2.6.27.7' make[1]: *** [modules] Error 2 make[1]: Leaving directory `/usr/src/zaptel-1.4.12.1' make: *** [all] Error 2 I believe that this is fixed in SVN of Zaptel. However you should note that the latest version of Zaptel is DAHDI. It has been fixed there as well. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/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] How to dial a number make two phone Ring at the same time?
On Thu, Jan 7, 2010 at 3:07 PM, Zhang Shukun bit...@gmail.com wrote: Thank you! but how can i determine whether ring at the same time or alternative ring? BTW, the uri http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.con It got mistyped or cut, it's http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf The concatenation I showed was for simultaneous ringing of devices. For the rest, yuou will be best served by looking through the docsz on dialplan and possibly queues. Best, /r ___ -- 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] Please remove me from the mailing list.
Can I be taken off the mailing list please. Thanks. rick ___ -- 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] Please remove me from the mailing list.
read your posting and it will tell you haw to remove yourself. On Thu, Jan 7, 2010 at 10:49 AM, Rick Dean ric.d...@gmail.com wrote: Can I be taken off the mailing list please. Thanks. rick ___ -- 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] Please remove me from the mailing list.
Go to this address for information on how to remove yourself:- http://lists.digium.com/mailman/listinfo/asterisk-users -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rick Dean Sent: 07 January 2010 15:50 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Please remove me from the mailing list. Can I be taken off the mailing list please. Thanks. rick ___ -- 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 smime.p7s Description: S/MIME cryptographic signature ___ -- 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] Sip REFER failes w/603 Decline (Policy), Polycom Phone
I have several sip stations that on a that are on a nat'd network behind a nice friend firewall.. no audio path issues, 2 way audio works, etc,etc,etc. However, I can't get any of my phones to Transfer or Blind Transfer.. I search and search, and well, just about gone nuts on this one. Here is sip debug from pressing transfer-blind-dial dest-Dial Key (note both stations do have access tot eh dial-dst ext of 202010) -- Started music on hold, class 'default', on channel 'SIP/1050-0a6ffa70' --- SIP read from XXX.XXX.232.66:8986 --- ACK sip:1...@xxx.xxx.232.175 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.232.66:8986;branch=z9hG4bK3dc3ce44DF7EE83D From: 1051 sip:1...@xxx.xxx.232.66:8986;tag=D117C080-6FFBC539 To: 1050 sip:1...@xxx.xxx.232.175;tag=as140f4415 CSeq: 1 ACK Call-ID: 75235b51626f63440c264f6b70dc5...@xxx.xxx.232.175 Contact: sip:1...@xxx.xxx.232.66:8986 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_331-UA/3.2.2.0477 Accept-Language: en Max-Forwards: 70 Content-Length: 0 - --- (12 headers 0 lines) --- --- SIP read from XXX.XXX.232.66:8986 --- REFER sip:1...@xxx.xxx.232.175 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.232.66:8986;branch=z9hG4bKd2064841D6E30B4A From: 1051 sip:1...@xxx.xxx.232.66:8986;tag=D117C080-6FFBC539 To: 1050 sip:1...@xxx.xxx.232.175;tag=as140f4415 CSeq: 2 REFER Call-ID: 75235b51626f63440c264f6b70dc5...@xxx.xxx.232.175 Contact: sip:1...@xxx.xxx.232.66:8986 User-Agent: PolycomSoundPointIP-SPIP_331-UA/3.2.2.0477 Accept-Language: en Refer-To: sip:202...@xxx.xxx.232.175;user=phone Referred-By: sip:1...@xxx.xxx.232.175 Max-Forwards: 70 Content-Length: 0 - --- (13 headers 0 lines) --- Call 75235b51626f63440c264f6b70dc5...@xxx.xxx.232.175 got a SIP call transfer from caller: (REFER)! --- Transmitting (no NAT) to XXX.XXX.232.66:8986 --- SIP/2.0 603 Declined (policy) Via: SIP/2.0/UDP XXX.XXX.232.66:8986;branch=z9hG4bKd2064841D6E30B4A;received=XXX.XXX.232.66 From: 1051 sip:1...@xxx.xxx.232.66:8986;tag=D117C080-6FFBC539 To: 1050 sip:1...@xxx.xxx.232.175;tag=as140f4415 Call-ID: 75235b51626f63440c264f6b70dc5...@xxx.xxx.232.175 CSeq: 2 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: sip:1...@xxx.xxx.232.175 Content-Length: 0 -- Stopped music on hold on SIP/1050-0a6ffa70 ___ -- 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] Sip REFER failes w/603 Decline (Policy), Polycom Phone
On Thu, Jan 7, 2010 at 11:15 AM, William Stillwell (Lists) william.stillwell-li...@ablebody.net wrote: I have several sip stations that on a that are on a nat'd network behind a nice friend firewall.. no audio path issues, 2 way audio works, etc,etc,etc. However, I can't get any of my phones to Transfer or Blind Transfer.. I search and search, and well, just about gone nuts on this one. Here is sip debug from pressing transfer-blind-dial dest-Dial Key (note both stations do have access tot eh dial-dst ext of 202010) -- Started music on hold, class 'default', on channel 'SIP/1050-0a6ffa70' --- SIP read from XXX.XXX.232.66:8986 --- ACK sip:1...@xxx.xxx.232.175 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.232.66:8986;branch=z9hG4bK3dc3ce44DF7EE83D From: 1051 sip:1...@xxx.xxx.232.66:8986;tag=D117C080-6FFBC539 To: 1050 sip:1...@xxx.xxx.232.175;tag=as140f4415 CSeq: 1 ACK Call-ID: 75235b51626f63440c264f6b70dc5...@xxx.xxx.232.175 Contact: sip:1...@xxx.xxx.232.66:8986 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_331-UA/3.2.2.0477 Accept-Language: en Max-Forwards: 70 Content-Length: 0 - --- (12 headers 0 lines) --- --- SIP read from XXX.XXX.232.66:8986 --- REFER sip:1...@xxx.xxx.232.175 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.232.66:8986;branch=z9hG4bKd2064841D6E30B4A From: 1051 sip:1...@xxx.xxx.232.66:8986;tag=D117C080-6FFBC539 To: 1050 sip:1...@xxx.xxx.232.175;tag=as140f4415 CSeq: 2 REFER Call-ID: 75235b51626f63440c264f6b70dc5...@xxx.xxx.232.175 Contact: sip:1...@xxx.xxx.232.66:8986 User-Agent: PolycomSoundPointIP-SPIP_331-UA/3.2.2.0477 Accept-Language: en Refer-To: sip:202...@xxx.xxx.232.175;user=phone Referred-By: sip:1...@xxx.xxx.232.175 Max-Forwards: 70 Content-Length: 0 - --- (13 headers 0 lines) --- Call 75235b51626f63440c264f6b70dc5...@xxx.xxx.232.175 got a SIP call transfer from caller: (REFER)! --- Transmitting (no NAT) to XXX.XXX.232.66:8986 --- SIP/2.0 603 Declined (policy) Via: SIP/2.0/UDP XXX.XXX.232.66:8986;branch=z9hG4bKd2064841D6E30B4A;received=XXX.XXX.232.66 From: 1051 sip:1...@xxx.xxx.232.66:8986;tag=D117C080-6FFBC539 To: 1050 sip:1...@xxx.xxx.232.175;tag=as140f4415 Call-ID: 75235b51626f63440c264f6b70dc5...@xxx.xxx.232.175 CSeq: 2 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: sip:1...@xxx.xxx.232.175 Content-Length: 0 -- Stopped music on hold on SIP/1050-0a6ffa70 Do you have notransfer=yes and canreinvite=no set anywhere? Just a shot in the dark. Thanks, Steve Totaro ___ -- 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] Sip REFER failes w/603 Decline (Policy), Polycom Phone
7 jan 2010 kl. 17.15 skrev William Stillwell (Lists): I have several sip stations that on a that are on a nat'd network behind a nice friend firewall.. no audio path issues, 2 way audio works, etc,etc,etc. However, I can't get any of my phones to Transfer or Blind Transfer.. I search and search, and well, just about gone nuts on this one. Check the allowtransfer setting in sip.conf. /Olle Here is sip debug from pressing transfer-blind-dial dest-Dial Key (note both stations do have access tot eh dial-dst ext of 202010) -- Started music on hold, class 'default', on channel 'SIP/1050-0a6ffa70' --- SIP read from XXX.XXX.232.66:8986 --- ACK sip:1...@xxx.xxx.232.175 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.232.66:8986;branch=z9hG4bK3dc3ce44DF7EE83D From: 1051 sip:1...@xxx.xxx.232.66:8986;tag=D117C080-6FFBC539 To: 1050 sip:1...@xxx.xxx.232.175;tag=as140f4415 CSeq: 1 ACK Call-ID: 75235b51626f63440c264f6b70dc5...@xxx.xxx.232.175 Contact: sip:1...@xxx.xxx.232.66:8986 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_331-UA/3.2.2.0477 Accept-Language: en Max-Forwards: 70 Content-Length: 0 - --- (12 headers 0 lines) --- --- SIP read from XXX.XXX.232.66:8986 --- REFER sip:1...@xxx.xxx.232.175 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.232.66:8986;branch=z9hG4bKd2064841D6E30B4A From: 1051 sip:1...@xxx.xxx.232.66:8986;tag=D117C080-6FFBC539 To: 1050 sip:1...@xxx.xxx.232.175;tag=as140f4415 CSeq: 2 REFER Call-ID: 75235b51626f63440c264f6b70dc5...@xxx.xxx.232.175 Contact: sip:1...@xxx.xxx.232.66:8986 User-Agent: PolycomSoundPointIP-SPIP_331-UA/3.2.2.0477 Accept-Language: en Refer-To: sip:202...@xxx.xxx.232.175;user=phone Referred-By: sip:1...@xxx.xxx.232.175 Max-Forwards: 70 Content-Length: 0 - --- (13 headers 0 lines) --- Call 75235b51626f63440c264f6b70dc5...@xxx.xxx.232.175 got a SIP call transfer from caller: (REFER)! --- Transmitting (no NAT) to XXX.XXX.232.66:8986 --- SIP/2.0 603 Declined (policy) Via: SIP/2.0/UDP XXX.XXX.232.66:8986;branch=z9hG4bKd2064841D6E30B4A;received=XXX.XXX.232.66 From: 1051 sip:1...@xxx.xxx.232.66:8986;tag=D117C080-6FFBC539 To: 1050 sip:1...@xxx.xxx.232.175;tag=as140f4415 Call-ID: 75235b51626f63440c264f6b70dc5...@xxx.xxx.232.175 CSeq: 2 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: sip:1...@xxx.xxx.232.175 Content-Length: 0 -- Stopped music on hold on SIP/1050-0a6ffa70 ___ -- 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 --- * Olle E Johansson - o...@edvina.net * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden ___ -- 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] Sip REFER failes w/603 Decline (Policy), Polycom Phone
Ok, im gonna go craw back under a rock.. Third line of my sip.conf allowtransfer=no Thanks for those who responded (Steve Ollie) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olle E. Johansson Sent: Thursday, January 07, 2010 11:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sip REFER failes w/603 Decline (Policy), Polycom Phone 7 jan 2010 kl. 17.15 skrev William Stillwell (Lists): I have several sip stations that on a that are on a nat'd network behind a nice friend firewall.. no audio path issues, 2 way audio works, etc,etc,etc. However, I can't get any of my phones to Transfer or Blind Transfer.. I search and search, and well, just about gone nuts on this one. Check the allowtransfer setting in sip.conf. /Olle Here is sip debug from pressing transfer-blind-dial dest-Dial Key (note both stations do have access tot eh dial-dst ext of 202010) -- Started music on hold, class 'default', on channel 'SIP/1050-0a6ffa70' --- SIP read from XXX.XXX.232.66:8986 --- ACK sip:1...@xxx.xxx.232.175 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.232.66:8986;branch=z9hG4bK3dc3ce44DF7EE83D From: 1051 sip:1...@xxx.xxx.232.66:8986;tag=D117C080-6FFBC539 To: 1050 sip:1...@xxx.xxx.232.175;tag=as140f4415 CSeq: 1 ACK Call-ID: 75235b51626f63440c264f6b70dc5...@xxx.xxx.232.175 Contact: sip:1...@xxx.xxx.232.66:8986 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_331-UA/3.2.2.0477 Accept-Language: en Max-Forwards: 70 Content-Length: 0 - --- (12 headers 0 lines) --- --- SIP read from XXX.XXX.232.66:8986 --- REFER sip:1...@xxx.xxx.232.175 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.232.66:8986;branch=z9hG4bKd2064841D6E30B4A From: 1051 sip:1...@xxx.xxx.232.66:8986;tag=D117C080-6FFBC539 To: 1050 sip:1...@xxx.xxx.232.175;tag=as140f4415 CSeq: 2 REFER Call-ID: 75235b51626f63440c264f6b70dc5...@xxx.xxx.232.175 Contact: sip:1...@xxx.xxx.232.66:8986 User-Agent: PolycomSoundPointIP-SPIP_331-UA/3.2.2.0477 Accept-Language: en Refer-To: sip:202...@xxx.xxx.232.175;user=phone Referred-By: sip:1...@xxx.xxx.232.175 Max-Forwards: 70 Content-Length: 0 - --- (13 headers 0 lines) --- Call 75235b51626f63440c264f6b70dc5...@xxx.xxx.232.175 got a SIP call transfer from caller: (REFER)! --- Transmitting (no NAT) to XXX.XXX.232.66:8986 --- SIP/2.0 603 Declined (policy) Via: SIP/2.0/UDP XXX.XXX.232.66:8986;branch=z9hG4bKd2064841D6E30B4A;received=XXX.XXX.232.66 From: 1051 sip:1...@xxx.xxx.232.66:8986;tag=D117C080-6FFBC539 To: 1050 sip:1...@xxx.xxx.232.175;tag=as140f4415 Call-ID: 75235b51626f63440c264f6b70dc5...@xxx.xxx.232.175 CSeq: 2 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: sip:1...@xxx.xxx.232.175 Content-Length: 0 -- Stopped music on hold on SIP/1050-0a6ffa70 ___ -- 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 --- * Olle E Johansson - o...@edvina.net * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden ___ -- 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] dns messages on console
Ever since upgrading to 1.6 I get messages like these. I want everything else that shows up, but is there a way to make all the dns messages go away? Ira doing dnsmgr_lookup for 'gw5.telasip.com' doing dnsmgr_lookup for 'sipconnect.ipcomms.net' doing dnsmgr_lookup for 'proxy.ideasip.com' ast_get_srv: SRV lookup for '_sip._UDP.proxy.ideasip.com' mapped to host proxy.ideasip.com, port 5060 ___ -- 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] Please remove me from the mailing list.
Steve Totaro wrote: read your posting and it will tell you haw to remove yourself. On Thu, Jan 7, 2010 at 10:49 AM, Rick Dean ric.d...@gmail.com mailto:ric.d...@gmail.com wrote: Can I be taken off the mailing list please. Thanks. rick http://lists.digium.com/mailman/listinfo/asterisk-users And a proper mail client will also parse the headers and provide unsubscribe information/buttons based on that... --FP ___ -- 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] Please remove me from the mailing list.
I've never seen that in Outlook. What client do you use? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Francesco Peeters Sent: 07 January 2010 18:58 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Please remove me from the mailing list. Steve Totaro wrote: read your posting and it will tell you haw to remove yourself. On Thu, Jan 7, 2010 at 10:49 AM, Rick Dean ric.d...@gmail.com mailto:ric.d...@gmail.com wrote: Can I be taken off the mailing list please. Thanks. rick http://lists.digium.com/mailman/listinfo/asterisk-users And a proper mail client will also parse the headers and provide unsubscribe information/buttons based on that... --FP ___ -- 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 smime.p7s Description: S/MIME cryptographic signature ___ -- 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] Please remove me from the mailing list.
Gmail DOES process those headers... And a proper mail client will also parse the headers and provide unsubscribe information/buttons based on that ___ -- 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] Please remove me from the mailing list.
Dan Journo wrote: I've never seen that in Outlook. What client do you use? Lately I have been using Thunderbird with an RFC2369 header plugin. --FP ___ -- 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] Please remove me from the mailing list.
I use gmail but don't see any buttons for unsubscribe or anything like that? Also, gmail defaults to top posting...which seems to upset some people 'round these parts. I have yet to find a way to make gmail not top-post by default... On Thu, Jan 7, 2010 at 1:16 PM, Francesco Peeters france...@fampeeters.comwrote: Dan Journo wrote: I've never seen that in Outlook. What client do you use? Lately I have been using Thunderbird with an RFC2369 header plugin. --FP ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, --Warren Selby http://www.selbytech.com ___ -- 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] Please remove me from the mailing list.
At 2:01 PM on 07 Jan 2010, Dan Journo wrote: I've never seen that in Outlook. What client do you use? Claws Mail provides a Mailing-List sub-menu under the Message menu, which includes Post, Subscribe and Unsubscribe options, among others. It's amazing what paying attention to standards can do for you... Steve Totaro wrote: read your posting and it will tell you haw to remove yourself. On Thu, Jan 7, 2010 at 10:49 AM, Rick Dean ric.d...@gmail.com mailto:ric.d...@gmail.com wrote: Can I be taken off the mailing list please. Thanks. rick http://lists.digium.com/mailman/listinfo/asterisk-users And a proper mail client will also parse the headers and provide unsubscribe information/buttons based on that... -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 signature.asc Description: PGP signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please remove me from the mailing list.
On Thu, Jan 7, 2010 at 1:27 PM, Warren Selby wcse...@selbytech.com wrote: I use gmail but don't see any buttons for unsubscribe or anything like that? Click on 'show details' at the top of the message and it will expand to show those options. I just found them over Christmas as I was trying to thin down the number of lists which flood my Inbox making things really simple. -Michael ___ -- 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] Please remove me from the mailing list.
On 7 Jan 2010, at 19:01, Dan Journo wrote: I've never seen that in Outlook. What client do you use? He said 'proper' mail client ;) *holy war* Sorry... S ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please remove me from the mailing list.
I haven't had a good mailing list war in a while. Yes, gmail DOES default to top posting, because bottom posting is silly (in general, but especially for a client that hides quoted text (like gmail)). Top posting is modern. And better. And doesn't make me scroll through 10 thousand messages and awful rsa keys to get to the message... FLAME AWAY!!! Press the 'show details' to the right hand side of the message box, then click the link that shows up that says 'unsubscribe'... -Dave snip I use gmail but don't see any buttons for unsubscribe or anything like that? Also, gmail defaults to top posting...which seems to upset some people 'round these parts. I have yet to find a way to make gmail not top-post by default... /snip ___ -- 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] Please remove me from the mailing list.
http://www.washington.edu/computing/mailman/faqs/mailman.email.html Em 07/01/2010, às 15:29, C. Chad Wallace cwall...@lodgingcompany.com escreveu: At 2:01 PM on 07 Jan 2010, Dan Journo wrote: I've never seen that in Outlook. What client do you use? Claws Mail provides a Mailing-List sub-menu under the Message menu, which includes Post, Subscribe and Unsubscribe options, among others. It's amazing what paying attention to standards can do for you... Steve Totaro wrote: read your posting and it will tell you haw to remove yourself. On Thu, Jan 7, 2010 at 10:49 AM, Rick Dean ric.d...@gmail.com mailto:ric.d...@gmail.com wrote: Can I be taken off the mailing list please. Thanks. rick http://lists.digium.com/mailman/listinfo/asterisk-users And a proper mail client will also parse the headers and provide unsubscribe information/buttons based on that... -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 ___ -- 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] Crash in Asterisk
My friends, I'm having some problems in my Asterisk, the thing is that Asterisk seem to be crashed (or dead) sometimes (2 times in 3 weeks) I noticed this today, when i could not make any internall call, tha calls to the voicemail (*1) did not work it just don't say nothing, nothing appears in console; i tried to make a CLIstop now but nothing happens, i could not stop the asterisk server The outgoing calls and incoming calls were also dead; seems that asterisk was not working but ït was up I had to reboot the server and now is better, working just fine... Here is some output from /var/log/asterisk/messages at the time of the Asterisk Crash [Jan 5 16:38:58] WARNING[6787] chan_sip.c: Autodestruct on dialog ' 4da606f808616e5379e299307824b...@10.4.1.6' with owner in place (Method: ACK) [Jan 5 16:39:06] WARNING[6787] chan_sip.c: Autodestruct on dialog ' 4da606f808616e5379e299307824b...@10.4.1.6' with owner in place (Method: BYE) [Jan 5 16:43:46] NOTICE[6787] chan_sip.c: Registration from ' sip:3...@10.4.1.6:5060' failed for '10.4.2.3' - No matching peer found [Jan 5 16:45:14] NOTICE[6787] chan_sip.c: Disconnecting call 'SIP/422-0a0b1e30' for lack of RTP activity in 301 seconds [Jan 5 16:49:21] NOTICE[6787] chan_sip.c: Peer '422' is now Reachable. (179ms / 2000ms) [Jan 5 16:51:08] NOTICE[6787] chan_sip.c: Peer '328' is now Reachable. (1ms / 2000ms) [Jan 5 16:51:19] WARNING[6787] channel.c: Channel allocation failed: Refusing due to active shutdown [Jan 5 16:51:19] WARNING[6787] chan_sip.c: Unable to allocate AST channel structure for SIP channel [Jan 5 16:51:19] NOTICE[6787] chan_sip.c: Unable to create/find SIP channel for this INVITE [Jan 5 16:51:54] ERROR[6787] res_config_mysql.c: MySQL RealTime: Ping failed (2003). Trying an explicit reconnect. [Jan 5 16:51:54] ERROR[6787] res_config_mysql.c: MySQL RealTime: Failed to connect database server dreampbx on 127.0.0.1 (err 2003). Check debug for more info. [Jan 5 16:51:54] ERROR[6787] res_config_mysql.c: MySQL RealTime: Failed to connect database server dreampbx on 127.0.0.1 (err 2003). Check debug for more info. [Jan 5 16:52:02] WARNING[6787] acl.c: Cannot connect [Jan 5 16:52:02] WARNING[6787] chan_sip.c: sip_xmit of 0xa151230 (len 510) to 10.4.2.3:5060 returned -2: Network is unreachable [Jan 5 16:52:06] NOTICE[6787] chan_sip.c: Peer '301' is now UNREACHABLE! Last qualify: 31 [Jan 5 16:57:39] NOTICE[5793] cdr.c: CDR simple logging enabled. [Jan 5 16:57:39] NOTICE[5793] loader.c: 150 modules will be loaded. [Jan 5 16:57:39] WARNING[5793] res_musiconhold.c: Cannot open dir /var/lib/asterisk/mohejemplo or dir does not exist [Jan 5 16:57:39] WARNING[5793] res_smdi.c: No SMDI interfaces are available to listen on, not starting SMDI listener. [Jan 5 16:57:40] WARNING[5793] pbx_dundi.c: Unable to look up host 'vetelcom' [Jan 5 16:57:40] NOTICE[5793] chan_ooh323.c: -- -- - --- *** IMPORTANT NOTE *** --- --- This module is currently unsupported. Use it at your own risk. --- What do you think my friends? How can i solve this problem? Thanks in advance DD ___ -- 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] Crash in Asterisk
On Thu, 7 Jan 2010 15:58:43 -0430 Danny Dias ing.diasda...@gmail.com wrote: My friends, I'm having some problems in my Asterisk, the thing is that Asterisk seem to be crashed (or dead) sometimes (2 times in 3 weeks) [Jan 5 16:51:19] WARNING[6787] channel.c: Channel allocation failed: Refusing due to active shutdown Hmm. Seems to start here. Are you sure someone isn't just restarting mysql? I'd find out what causes this string to be issued in channel.c, no? [Jan 5 16:51:54] ERROR[6787] res_config_mysql.c: MySQL RealTime: Ping failed (2003). Trying an explicit reconnect. But that's pretty obviously mysql unavailable. [Jan 5 16:51:54] ERROR[6787] res_config_mysql.c: MySQL RealTime: Failed to connect database server dreampbx on 127.0.0.1 (err 2003). Check debug for more info. mysql on the same machine, that is? So have a look there? [Jan 5 16:51:54] ERROR[6787] res_config_mysql.c: MySQL RealTime: Failed to connect database server dreampbx on 127.0.0.1 (err 2003). Check debug for more info. What do you think my friends? How can i solve this problem? Find out why mysql isn't talking to asterisk, or don't use mysql realtime? Just my guess. Good luck, -- Michael Higgins ___ -- 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] Faxing: Anyone have a compiled executable?
On Thu, Jan 07, 2010 at 12:50:03AM -0600, Doug wrote: At 00:22 1/7/2010, Tzafrir Cohen wrote: On Wed, Jan 06, 2010 at 11:41:54PM -0600, Doug wrote: At 16:49 1/5/2010, Tzafrir Cohen wrote: On Tue, Jan 05, 2010 at 04:24:37PM -0600, Doug wrote: Hi, Having problems with getting either RxFax or FaxReceive to compile. Running Asterisk 1.4 on CentOS 5. What version of SpanDSP do you use? spandsp-0.0.6pre12.tgz and: libtiff-3.8.2-7.el5_3.4 libtiff-devel-3.8.2-7.el5_3.4 Which do you recommend? What errors do you get? I'm using a backport of app_fax.c and it works well. Do you have the link for the C source? app_fax.c from: https://agx-ast-addons.svn.sourceforge.net/svnroot/agx-ast-addons/trunk/app-spandsp/ Just remove the '#include ../addon_version.h line, and the single include used from it (AGX_AST_ADDON_VERSION). -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/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] Faxing: Anyone have a compiled executable?
Has there been any improvement with app_fax ? I stopped using it as I had a high failure rate with inbound faxes (10%+) 1000 faxes a week ,with over a 100 failures can get quite annoying from people complaining.. I could get it to fail everytime I tried sending a solid black fax page. (ie, take a sheet of paper that is all black, or heavily black, and fax it, I got a ton of errors, or just plain rx reception failure).. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Thursday, January 07, 2010 4:33 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Faxing: Anyone have a compiled executable? On Thu, Jan 07, 2010 at 12:50:03AM -0600, Doug wrote: At 00:22 1/7/2010, Tzafrir Cohen wrote: On Wed, Jan 06, 2010 at 11:41:54PM -0600, Doug wrote: At 16:49 1/5/2010, Tzafrir Cohen wrote: On Tue, Jan 05, 2010 at 04:24:37PM -0600, Doug wrote: Hi, Having problems with getting either RxFax or FaxReceive to compile. Running Asterisk 1.4 on CentOS 5. What version of SpanDSP do you use? spandsp-0.0.6pre12.tgz and: libtiff-3.8.2-7.el5_3.4 libtiff-devel-3.8.2-7.el5_3.4 Which do you recommend? What errors do you get? I'm using a backport of app_fax.c and it works well. Do you have the link for the C source? app_fax.c from: https://agx-ast-addons.svn.sourceforge.net/svnroot/agx-ast-addons/trunk/app- spandsp/ Just remove the '#include ../addon_version.h line, and the single include used from it (AGX_AST_ADDON_VERSION). -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/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 ___ -- 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] voicemail /odbc problem
Hi, I'm having a bit of a problem with storing voicemail messages in an odbc database. I *think* I've got everything configured correctly but messages are stored on the asterisk server instread of in the database. System info 64 bit redhat RHEL 5.1 Asterisk 1.4.26 unixODBC installed used makemenuselect to instal res_odbc and cdr_odbc Back end database DB2 Database name voiceml Tables created on server AST_CDR AST_CONFIG VOICEMESSAGES I've checked the unix ODBC side of things and I can connect to the DB2 database o.k. I've set up cdr_odbc.conf [global] dsn=voiceml username=usernme password=password loguniqueid=yes dispositionstring=yes table=ast_cdr ;cdr is default table name usegmtime=yes ; set to yes to log in GMT res_odbc.conf odbcstorage = voiceml [voiceml] enabled = yes dsn = voiceml username = userid password = password pre-connect = yes in extconfig.conf I've got [settings] ; voicemail = odbc,voiceml musiconhold.conf = odbc,voiceml,ast_config I've set up an IAX2 test user called [odbc_test_user] type=friend secret=supersecret context=odbc_vm_test host=dynamic qualify=yes disallow=all allow=ulaw allow=gsm and a context of [odbc_vm_test] exten = 100,1,Voicemail(88...@default);leave a message exten = 200,1,VoicemailMain(88...@default) ; retrieve a mail message When I connect from my MAC using zoiper and call extension 100 I can leave a voicemail message. Looking in /var/log/asterisk/full I get [Jan 7 21:58:34] DEBUG[7257] pbx.c: Launching 'VoiceMail' [Jan 7 21:58:34] VERBOSE[7257] logger.c: -- Executing [...@odbc_vm_test:1] VoiceMail(IAX2/odbc_test_use r-8660, 88...@default) in new stack [Jan 7 21:58:34] DEBUG[7257] chan_iax2.c: Answering IAX2 call [Jan 7 21:58:34] DEBUG[7257] devicestate.c: Notification of state change to be queued on device/channel IAX2 /odbc_test_user [Jan 7 21:58:34] DEBUG[7204] chan_iax2.c: Checking device state for device odbc_test_user [Jan 7 21:58:34] DEBUG[7204] chan_iax2.c: iax2_devicestate: Found peer. What's device state of odbc_test_use r? addr=57011606, defaddr=0 maxms=2000, lastms=41 [Jan 7 21:58:34] DEBUG[7204] devicestate.c: Changing state for IAX2/ odbc_test_user - state 2 (In use) [Jan 7 21:58:34] DEBUG[7257] app_voicemail.c: Before find_user [Jan 7 21:58:34] DEBUG[7230] app_queue.c: Device 'IAX2/ odbc_test_user' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jan 7 21:58:34] VERBOSE[7257] logger.c: -- IAX2/ odbc_test_user-8660 Playing 'vm-intro' (language 'en' ) [Jan 7 21:58:34] DEBUG[7234] chan_iax2.c: Ooh, voice format changed to 4 [Jan 7 21:58:40] DEBUG[7257] app.c: Locked path '/var/spool/asterisk/ voicemail/default/8/INBOX' [Jan 7 21:58:40] DEBUG[7257] app.c: Unlocked path '/var/spool/ asterisk/voicemail/default/8/INBOX' [Jan 7 21:58:40] VERBOSE[7257] logger.c: -- IAX2/ odbc_test_user-8660 Playing 'beep' (language 'en') [Jan 7 21:58:40] VERBOSE[7257] logger.c: -- Recording the message [Jan 7 21:58:40] DEBUG[7257] app.c: play_and_record: None, /var/ spool/asterisk/voicemail/default/8/tmp /Ob6NSJ, 'wav49' [Jan 7 21:58:40] DEBUG[7257] app.c: Recording Formats: sfmts=wav49 [Jan 7 21:58:40] VERBOSE[7257] logger.c: -- x=0, open writing: / var/spool/asterisk/voicemail/default/88 888/tmp/Ob6NSJ format: wav49, 0x1c1bd0f8 [Jan 7 21:58:40] DEBUG[7240] chan_iax2.c: Peer odbc_test_user: got pong, lastms 41, historicms 41, maxms 200 0 [Jan 7 21:59:12] VERBOSE[7257] logger.c: -- User hung up [Jan 7 21:59:12] DEBUG[7257] app.c: Locked path '/var/spool/asterisk/ voicemail/default/8/INBOX' [Jan 7 21:59:12] DEBUG[7257] app.c: Unlocked path '/var/spool/ asterisk/voicemail/default/8/INBOX' user-8660' 2/odbc_test_user-8660' [Jan 7 21:59:12] DEBUG[7257] channel.c: Soft-Hanging up channel 'IAX2/ odbc_test_user-8660' [Jan 7 21:59:12] DEBUG[7257] channel.c: Hanging up channel 'IAX2/ odbc_test_user-8660' [Jan 7 21:59:12] DEBUG[7257] chan_iax2.c: We're hanging up IAX2/ odbc_test_user-8660 now... [Jan 7 21:59:12] DEBUG[7257] chan_iax2.c: Really destroying IAX2/ odbc_test_user-8660 now... [Jan 7 21:59:12] VERBOSE[7257] logger.c: -- Hungup 'IAX2/ odbc_test_user-8660' /odbc_test_user I can call the asterisk server an listen to the voicemail message but there's nothing in the database voicemessages table. I've enabled unixODBC logging and although I can see log entries accessing the AST_CONFIG and AST_CDR tables, there's nothing accessing the voicemessages table. Any idea where I can look next? TIA Alex Checked by Hu-fw-yhman ___ -- 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] Faxing: Anyone have a compiled executable?
On Thu, Jan 07, 2010 at 05:05:09PM -0500, William Stillwell (Lists) wrote: Has there been any improvement with app_fax ? Builds with spandsp 0.0.6 (as opposed to older versions that required older versions of spandsp). It is also a more well-behaving Asterisk app in its logging (does not keep its own personal log). -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/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
[asterisk-users] AGI perl script set timeout within script?
Hi All, I'm running an AGI, calling a perl script the does number lookups to a remote server. I would like to put a timeout in the script. The problem I'm running into is if the DNS server is not responding, the script hangs and waits for 30 seconds before returning to the Asterisk dialplan. I would like a timeout of 1 second, then return. Here is my clean script: *** #!/usr/bin/perl $|=1; #Modules to Use ### use Asterisk::AGI; $AGI = new Asterisk::AGI; my %input = $AGI-ReadParse(); # Set variables according to supplied arguments $number = $ARGV[0]; $AGI-exec(agi,agi://agi.server.com/script.agi?user=usernamenumber=$number); *** Any assistance will be appreciated. Thanks. JR -- JR Richardson Engineering for the Masses ___ -- 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] AGI perl script set timeout within script?
On Thu, 7 Jan 2010, JR Richardson wrote:
I'm running an AGI, calling a perl script the does number lookups to a
remote server. I would like to put a timeout in the script. The
problem I'm running into is if the DNS server is not responding, the
script hangs and waits for 30 seconds before returning to the Asterisk
dialplan. I would like a timeout of 1 second, then return.
I'm a C weenie, so I can't provide Perl code. But it should look something
like this:
// handle the alarm
static voidlookup_failed
(
void
)
{
exit(EXIT_FAILURE);
}
int main
(
)
...
// set a signal alarm handler
signal(SIGALRM, (void (*)(int))(int)lookup_failed);
// set the alarm to go off in 1 second
alarm(1);
// lookup my number
do_lookup(dnis);
// cancel the alarm
alarm(0);
...
Of course, solving the real issue (DNS lookups), or masking it with a
local caching server or even a /etc/hosts file are viable alternatives.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Realtime LDAP Queues crashes
What are the LDAP searches like?
On 05/01/2010, Jorge Salamero Sanz ben...@cauterized.net wrote:
Hi all,
I've updated Asterisk trunk LDAP schema [0] [1] to include queues and other
attributes needed for a working LDAP backend (I'll open a bug to include
these
changes on svn).
SIP users and dialplan are perfectly working, but when I call a queue the
whole Asterisk (1.6.2.0) crashes:
on extconfig:
[settings]
sipusers = ldap,dc=nodomain,sip
sippeers = ldap,dc=nodomain,sip
extensions = ldap,dc=nodomain,extensions
voicemail = ldap,dc=nodomain,voicemail
queue_members = ldap,dc=nodomain,queue_member
queues = ldap,dc=nodomain,queue
on res_ldap.conf: see [1]
for the Queues on LDAP I have:
ou=Queues,dc=nodomain
ou: Queues
objectClass: top
objectClass: organizationalUnit
cn=foobar,ou=Queues,dc=nodomain
objectClass: applicationProcess
objectClass: AsteriskQueue
AstQueueName: foobar
AstQueueContext: default
AstQueueTimeout: 180
cn: foobar
the dialplan (on extensions.conf, the same if it's on LDAP):
[frontdesk]
exten = 78,1,Answer
exten = 78,n,Queue(foobar)
exten = 78,n,Hangup
[default]
include = common
include = frontdesk
switch = Realtime
and the user on LDAP:
uid=foo,ou=Users,dc=nodomain
cn: foo foo
uid: foo
sn: foo
uidNumber: 2002
gidNumber: 1901
homeDirectory: /nonexistent
userPassword: {SHA}C+7Hteo/D9vJXQ3UfzxbwnXaijM=
eboxSha1Password: {SHA}C+7Hteo/D9vJXQ3UfzxbwnXaijM=
eboxMd5Password: {MD5}rL0Y20zC+Fzt72VPzMSk2A==
eboxLmPassword: 5BFAFBEBFB6A0942AAD3B435B51404EE
eboxNtPassword: AC8E657F83DF82BEEA5D43BDAF7800CC
eboxDigestPassword: {MD5}x0Z+Prb70OIF3iARsuJ3Xg==
eboxRealmPassword: {MD5}c7467e3eb6fbd0e205de2011b2e2775e
givenName: foo
description: foo
AstAccountType: friend
AstAccountContext: users
AstAccountCallerID: 1001
AstAccountMailbox: 1001
AstAccountHost: dynamic
AstAccountNAT: yes
AstAccountQualify: yes
AstAccountCanReinvite: no
AstAccountDTMFMode: rfc2833
AstAccountInsecure: port
AstAccountLastQualifyMilliseconds: 0
AstAccountIPAddress: 0.0.0.0
AstAccountPort: 0
AstAccountExpirationTimestamp: 0
AstAccountRegistrationServer: 0
AstAccountUserAgent: 0
AstAccountFullContact: sip:0.0.0.0
AstContext: users
AstVoicemailMailbox: 1001
AstVoicemailPassword: 1001
AstVoicemailEmail: u...@domain
AstVoicemailAttach: yes
AstVoicemailDelete: no
AstQueueMembername: foobar
AstQueueMemberof: foobar
objectClass: AsteriskQueueMember
objectClass: AsteriskSIPUser
objectClass: AsteriskVoiceMail
objectClass: inetOrgPerson
objectClass: passwordHolder
objectClass: posixAccount
AstQueueInterface: SIP/1001
when i call the queue extension, on slapd I can see how Asterisk fetches the
AsteriskQueue objectClass, and then fetches the foo user, but then crashes
like this:
-- Executing [...@users:1] Answer(SIP/demo-, ) in new stack
-- Executing [...@users:2] Queue(SIP/demo-, foobar) in new
stack
[Jan 5 13:26:28] WARNING[6195]: app_queue.c:1134 create_queue_member: No
location at interface ''
[1]6124 segmentation fault (core dumped) asterisk -
vvc
*CLI queue show foobar
[1]6356 segmentation fault (core dumped) asterisk -
vvc
*CLI queue add member SIP/foo to foobar
[1]6394 segmentation fault (core dumped) asterisk -
vvc
any clue on what's wrong ? how could i debug this ? maybe there is some
attribute missing ? or the LDAP schema is wrong ? anyone with a working
setup
like this ?
thanks in advance !
[0] http://people.ebox-platform.com/~bencer/asterisk.ldif
[1] http://people.ebox-platform.com/~bencer/res_ldap.conf.mas
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Re: [asterisk-users] AGI perl script set timeout within script?
On Thu, Jan 7, 2010 at 6:27 PM, JR Richardson jmr.richard...@gmail.com wrote: problem I'm running into is if the DNS server is not responding, the script hangs and waits for 30 seconds before returning to the Asterisk dialplan. I would like a timeout of 1 second, then return. A few things... * stop using DNS? Problem solved. * put nagios monitoring on your DNS server? * put in a second DNS server, and tune your DNS timeout to a very low value in /etc/resolv.conf (read the man page) before jumping to next server? Or you could use the Perl language feature, which is called 'alarm'. Google around for some code samples. None of these are actually specific to asterisk, as it turns out. I don't know of any explicit asterisk method to force a timeout. ___ -- 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] voicemail /odbc problem
On Thursday 07 January 2010 16:20:38 Alex Sharaz wrote: I'm having a bit of a problem with storing voicemail messages in an odbc database. I *think* I've got everything configured correctly but messages are stored on the asterisk server instread of in the database. System info 64 bit redhat RHEL 5.1 Asterisk 1.4.26 unixODBC installed used makemenuselect to instal res_odbc and cdr_odbc Voicemail is kind of tricky. While it does use realtime for the storage of user mailbox settings, it does not use realtime for the storage of messages themselves. To do this, you'll need to run 'make menuselect' in the build directory, then, in the Voicemail Options section, enable ODBC_STORAGE, then recompile and reinstall (only the app_voicemail.so and app_directory.so modules). You'll also need to configure the odbc storage mechanism in voicemail.conf, specifically odbcstorage and odbctable. I agree that this is somewhat non-intuitive, so we may change this in the future to make it configurable at runtime, rather than an option at buildtime. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing: Anyone have a compiled executable?
On 01/08/2010 06:05 AM, William Stillwell (Lists) wrote: Has there been any improvement with app_fax ? I stopped using it as I had a high failure rate with inbound faxes (10%+) 1000 faxes a week ,with over a 100 failures can get quite annoying from people complaining.. I could get it to fail everytime I tried sending a solid black fax page. (ie, take a sheet of paper that is all black, or heavily black, and fax it, I got a ton of errors, or just plain rx reception failure).. If you want an idea of the performance level to expect from app_fax see http://www.soft-switch.org/spandsp-soft-fax-performance.html If 10% of your FAXes are failing, and they really have the potential to succeed (i.e. not voice calls, wrong numbers, etc), that's awful. Anything above 1% is poor. app_fax on a well set up system, with no timing issues, achieves 99% success for PSTN calls. Results with calls on the internet will vary, depending on the quality of your VoIP links. T.38 calls are generally far more reliable than audio ones across the internet. Are you using PSTN or VoIP connexions? Sending a black page is no harder than sending a white one. If you want a real stress test, try the checkerboard pattern TIFF file page, amongst the spandsp test data. That takes about half an hour to send one page. On a well set up system you should be able to send or receive those pages all day. If you can't, you probably have timing issues in your Asterisk setup. If you are going to ask if something has improved, its rather important to say which versions you are running now, and how you use them. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI perl script set timeout within script?
On Thursday 07 January 2010 18:59:24 David Backeberg wrote: On Thu, Jan 7, 2010 at 6:27 PM, JR Richardson jmr.richard...@gmail.com wrote: problem I'm running into is if the DNS server is not responding, the script hangs and waits for 30 seconds before returning to the Asterisk dialplan. I would like a timeout of 1 second, then return. A few things... * stop using DNS? Problem solved. * put nagios monitoring on your DNS server? * put in a second DNS server, and tune your DNS timeout to a very low value in /etc/resolv.conf (read the man page) before jumping to next server? Or you could use the Perl language feature, which is called 'alarm'. Google around for some code samples. Ah, but Perl isn't actually doing the DNS lookup. If you examine his script, he's merely passing back a name to the Asterisk process, which is then calling inet_aton(), which is the reason why he cannot control it from within the script. What he'd actually need to do is to start using Net::DNS to do the resolution on that name, first, perhaps even going as far as to connect to the server himself, and relay the channel between the AGI interface and the remote TCP interface. Then, he could use alarm() or the Time::Hires module to ensure his own timeouts override the builtins. But as it stands now, it's all Asterisk. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI perl script set timeout within script?
On Thu, Jan 7, 2010 at 6:27 PM, JR Richardson jmr.richard...@gmail.com wrote: problem I'm running into is if the DNS server is not responding, the script hangs and waits for 30 seconds before returning to the Asterisk dialplan. ?I would like a timeout of 1 second, then return. On Thursday 07 January 2010 18:59:24 David Backeberg wrote: * stop using DNS? Problem solved. * put nagios monitoring on your DNS server? * put in a second DNS server, and tune your DNS timeout to a very low value in /etc/resolv.conf (read the man page) before jumping to next server? Or you could use the Perl language feature, which is called 'alarm'. Google around for some code samples. On Thu, 7 Jan 2010, Tilghman Lesher wrote: Ah, but Perl isn't actually doing the DNS lookup. If you examine his script, he's merely passing back a name to the Asterisk process, which is then calling inet_aton(), which is the reason why he cannot control it from within the script. What he'd actually need to do is to start using Net::DNS to do the resolution on that name, first, perhaps even going as far as to connect to the server himself, and relay the channel between the AGI interface and the remote TCP interface. Then, he could use alarm() or the Time::Hires module to ensure his own timeouts override the builtins. But as it stands now, it's all Asterisk. If the DNS lookup is being done by Asterisk to resolve the FastAGI server name. If the DNS lookup is for the (assumed) database server in his script then the suggestions to use alarm() would do the trick. I guess we need clarification from the OP. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please remove me from the mailing list.
Careful, or Steve will un top post YOU! David Gibbons wrote: I haven’t had a good mailing list war in a while. Yes, gmail DOES default to top posting, because bottom posting is silly (in general, but especially for a client that hides quoted text (like gmail)). Top posting is modern. And better. And doesn’t make me scroll through 10 thousand messages and awful rsa keys to get to the message… FLAME AWAY!!! Press the ‘show details’ to the right hand side of the message box, then click the link that shows up that says ‘unsubscribe’… -Dave snip I use gmail but don't see any buttons for unsubscribe or anything like that? Also, gmail defaults to top posting...which seems to upset some people 'round these parts. I have yet to find a way to make gmail not top-post by default... /snip ___ -- 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 Checked by AVG - www.avg.com Version: 9.0.725 / Virus Database: 270.14.128/2604 - Release Date: 01/06/10 14:35:00 -- Dog is my co-pilot ___ -- 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] AGI perl script set timeout within script?
On Thu, Jan 7, 2010 at 6:27 PM, JR Richardson
jmr.richard...@gmail.com
wrote:
problem I'm running into is if the DNS server is not responding, the
script hangs and waits for 30 seconds before returning to the Asterisk
dialplan. ?I would like a timeout of 1 second, then return.
On Thursday 07 January 2010 18:59:24 David Backeberg wrote:
* stop using DNS? Problem solved. * put nagios monitoring on your DNS
server? * put in a second DNS server, and tune your DNS timeout to a
very low value in /etc/resolv.conf (read the man page) before jumping
to next server?
Or you could use the Perl language feature, which is called 'alarm'.
Google around for some code samples.
On Thu, 7 Jan 2010, Tilghman Lesher wrote:
Ah, but Perl isn't actually doing the DNS lookup. If you examine his
script, he's merely passing back a name to the Asterisk process, which
is then calling inet_aton(), which is the reason why he cannot control
it from within the script. What he'd actually need to do is to start
using Net::DNS to do the resolution on that name, first, perhaps even
going as far as to connect to the server himself, and relay the channel
between the AGI interface and the remote TCP interface.
Then, he could use alarm() or the Time::Hires module to ensure his own
timeouts override the builtins. But as it stands now, it's all
Asterisk.
If the DNS lookup is being done by Asterisk to resolve the FastAGI server
name. If the DNS lookup is for the (assumed) database server in his script
then the suggestions to use alarm() would do the trick.
I guess we need clarification from the OP.
I tend to agree with Tilghman on this. I tried the perl script eval, alarm,
$SIG{ALRM} functions till I was blue in the face from cussing at the screen.
It does not appear that the perl script is doing the DNS query, otherwise
the eval alarm would timeout and pass control back to asterisk. Another
indication is that '#define MAX_AGI_CONNECT 2000' in res_agi is not being
invoked because the timeout is around 30 seconds. Is that 30 second timeout
built into Asterisk? Can I put an absolute timeout on an agi script from
the dialplan prior to calling the agi application? Maybe I'll fork a macro
with a timeout, yea, that's it, let start forking, something new to cuss at.
Thanks for your input guys.
JR
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Re: [asterisk-users] AGI perl script set timeout within script?
On Thu, Jan 7, 2010 at 6:27 PM, JR Richardson
jmr.richard...@gmail.com
wrote:
problem I'm running into is if the DNS server is not responding, the
script hangs and waits for 30 seconds before returning to the
Asterisk dialplan. ?I would like a timeout of 1 second, then return.
On Thu, 7 Jan 2010, JR Richardson wrote:
I tried the perl script eval, alarm, $SIG{ALRM} functions till I was
blue in the face from cussing at the screen. It does not appear that the
perl script is doing the DNS query, otherwise the eval alarm would
timeout and pass control back to asterisk. Another indication is that
'#define MAX_AGI_CONNECT 2000' in res_agi is not being invoked because
the timeout is around 30 seconds. Is that 30 second timeout built into
Asterisk? Can I put an absolute timeout on an agi script from the
dialplan prior to calling the agi application? Maybe I'll fork a macro
with a timeout, yea, that's it, let start forking, something new to cuss
at.
What about:
1) Fixing the slow responding DNS server?
2) Tweaking /etc/resolv.conf options?
3) Setting up a caching name server on your Asterisk host?
4) Adding the AGI server host name and IP address to /etc/hosts?
5) Using the IP address of the AGI server in your dialplan?
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] AGI perl script set timeout within script?
On Thursday 07 January 2010 21:17:52 JR Richardson wrote:
On Thu, 7 Jan 2010, Tilghman Lesher wrote:
On Thu, Jan 7, 2010 at 6:27 PM, JR Richardson wrote:
problem I'm running into is if the DNS server is not responding, the
script hangs and waits for 30 seconds before returning to the
Asterisk dialplan. ?I would like a timeout of 1 second, then return.
Ah, but Perl isn't actually doing the DNS lookup. If you examine his
script, he's merely passing back a name to the Asterisk process, which
is then calling inet_aton(), which is the reason why he cannot control
it from within the script. What he'd actually need to do is to start
using Net::DNS to do the resolution on that name, first, perhaps even
going as far as to connect to the server himself, and relay the channel
between the AGI interface and the remote TCP interface.
Then, he could use alarm() or the Time::Hires module to ensure his own
timeouts override the builtins. But as it stands now, it's all
Asterisk.
I tried the perl script eval, alarm,
$SIG{ALRM} functions till I was blue in the face from cussing at the
screen. It does not appear that the perl script is doing the DNS query,
otherwise the eval alarm would timeout and pass control back to asterisk.
Another indication is that '#define MAX_AGI_CONNECT 2000' in res_agi is not
being invoked because the timeout is around 30 seconds. Is that 30 second
timeout built into Asterisk? Can I put an absolute timeout on an agi
script from the dialplan prior to calling the agi application? Maybe I'll
fork a macro with a timeout, yea, that's it, let start forking, something
new to cuss at.
No, the timeout is built into glibc. I don't see any documented method for
altering it, sorry. The only way to really do it in a way where you can
control the timeouts would be to do it in your Perl script, in the way that I
described above.
--
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com www.asterisk.org
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