[asterisk-users] Removing `chan_dahdi.conf`
Hi guys, I'm currently playing around with optimizing an Asterisk install (trying to remove as many possible configuration files as possible) for testing and debugging purposes. I've been able to remove most of the files and maintain an error-less Asterisk full file, with a single exception: I can't seem to remove `chan_dahdi.conf` without having Asterisk complain in the full logs: ERROR[1182] chan_dahdi.c: Unable to load config chan_dahdi.conf I'm only using DAHDI for dahdi_dummy timing, and nothing else. If I have an empty `chan_dahdi.conf` file, then Asterisk won't complain at all, but if I remove the empty file, Asterisk spits out the above error message. So my questions are: 1. By removing `chan_dahdi.conf`, am I breaking any functionality that would cause dahdi_dummy to not provide timing (or some other critical feature) in my Asterisk environment? 2. Is there someway to let Asterisk know that I *only* need DAHDI for the dahdi_dummy driver, and that the configuration file is unnecessary? (This way I don't get errors in my logs.) The point of me pursuing this is because it *seems* as if I can currently `get by` without having the `chan_dahdi.conf` file at all, even with the error message I'm getting in the logs. I just worry that there may be another consequence for not having this file that I will experience at some point, and since I haven't found one yet, I wanted to check with the experts to get a definitive answer. I'm currently using Asterisk 1.6.1.1. I plan to go through the source code for the module loader to see what I can find, but I figured I'd pop this email off anyhow to see what you all have to say. Thanks for your time. -- Randall Degges *http://rdegges.com/* -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realtime sip peers : musiconhold class
I took this from the wiki, but it's not working : [r...@asterisk testing]# sox test.wav -r 8000 -c1 test.alaw resample -ql sox sox: effect `resample' is deprecated; see sox(1) for an alternative sox formats: no handler for file extension `alaw' Jonas. On 08/14/2010 04:30 PM, Motiejus Jakštys wrote: On Sat, Aug 14, 2010 at 4:24 PM, Jonas Kellens jonas.kell...@telenet.be mailto:jonas.kell...@telenet.be wrote: intro extended version.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 44100 Hz You need *MONO, 8000Hz* $ man sox -- Motiejus Jakštys -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realtime sip peers : musiconhold class
And even when I think the format is correct : [r...@asterisk testing]# sox test.wav -r 8000 -c1 test.wav resample -ql sox sox: effect `resample' is deprecated; see sox(1) for an alternative sox wav: Premature EOF on .wav input file [r...@asterisk testing]# file test.wav test.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz It is still not working : [r...@asterisk testing]# asterisk -rx file convert test.wav test.alaw Unable to open input file: test.wav Jonas. On 08/14/2010 04:30 PM, Motiejus Jakštys wrote: On Sat, Aug 14, 2010 at 4:24 PM, Jonas Kellens jonas.kell...@telenet.be mailto:jonas.kell...@telenet.be wrote: intro extended version.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 44100 Hz You need *MONO, 8000Hz* $ man sox -- Motiejus Jakštys -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Removing `chan_dahdi.conf`
On Sat, Aug 14, 2010 at 11:46:49PM -0700, Randall Degges wrote: Hi guys, I'm currently playing around with optimizing an Asterisk install (trying to remove as many possible configuration files as possible) for testing and debugging purposes. I've been able to remove most of the files and maintain an error-less Asterisk full file, with a single exception: I can't seem to remove `chan_dahdi.conf` without having Asterisk complain in the full logs: ERROR[1182] chan_dahdi.c: Unable to load config chan_dahdi.conf I'm only using DAHDI for dahdi_dummy timing, and nothing else. If I have an empty `chan_dahdi.conf` file, then Asterisk won't complain at all, but if I remove the empty file, Asterisk spits out the above error message. So my questions are: 1. By removing `chan_dahdi.conf`, am I breaking any functionality that would cause dahdi_dummy to not provide timing (or some other critical feature) in my Asterisk environment? No. chan_dahdi is not required for using DAHDI as a timing source. 2. Is there someway to let Asterisk know that I *only* need DAHDI for the dahdi_dummy driver, and that the configuration file is unnecessary? (This way I don't get errors in my logs.) If you don't need chan_dahdi.so, why do you load it? Look into 'noload' in modules.conf . Consider not loading modules automatically. Consider removing modules you don't need from the modules directory (Consider. Don't just do without considering). There are in fact a number of other modules you don't need. Specifically other channel drivers (chan_*). Also look at the various res_* (do you need mysql/pgsql/odbc/sqlite/whatever? I bet you don't use all of those together ;-) The point of me pursuing this is because it *seems* as if I can currently `get by` without having the `chan_dahdi.conf` file at all, even with the error message I'm getting in the logs. I just worry that there may be another consequence for not having this file that I will experience at some point, and since I haven't found one yet, I wanted to check with the experts to get a definitive answer. I'm currently using Asterisk 1.6.1.1. Any reason you don't use a newer version (1.6.2.x? A newer 1.6.1.x?) -- 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Timing on Asterisk
Hi All I am occasionally hearing a slight pop or skip in an audio message playback on one ALL SIP installation. There are other Audio problems with the installation too (underwater Audio, 1 second 1 way audio delay (takes 1 second for Audio spoken by the customer to reach the agent, and Robotic voices). I have looked up many guides and memorized voip-info.org... I am starting to wonder if the problem is in my timing. I was using dahdi_dummy, I have put in a sangoma card and I am using it as a timing source (and only as a timing source). Here is the output of my timing source below. *CLI timing test 50 Attempting to test a timer with 50 ticks per second. Using the 'DAHDI' timing module for this test. It has been 1002 milliseconds, and we got 51 timer ticks *CLI timing test 100 Attempting to test a timer with 100 ticks per second. Using the 'DAHDI' timing module for this test. It has been 1002 milliseconds, and we got 102 timer ticks *CLI timing test 500 Attempting to test a timer with 500 ticks per second. Using the 'DAHDI' timing module for this test. It has been 1001 milliseconds, and we got 510 timer ticks *CLI timing test 10 Attempting to test a timer with 10 ticks per second. Using the 'DAHDI' timing module for this test. It has been 1080 milliseconds, and we got 11 timer ticks in asterisk.conf, internal_timing = yes -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 603 error
Hi, I have an interesting problem that the dial out via sip always generates 603 error The following is the sip debug Your help is appreciated. CK == Using SIP RTP CoS mark 5 -- Executing [998560...@dlpn_dp1:1] Dial(SIP/6100-005b, SIP/13398560...@hkbn2b) in new stack == Using SIP RTP CoS mark 5 Audio is at 113.253.230.26 port 11316 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 203.80.89.139:5060: INVITE sip:13398560...@s2hkbntel.net:5060 SIP/2.0 Via: SIP/2.0/UDP 113.253.230.26:5060;branch=z9hG4bK575022bd;rport Max-Forwards: 70 From: ck...@mobile sip:3594410...@s2hkbntel.netsip%3a3594410...@s2hkbntel.net ;tag=as1d554c43 To: sip:13398560...@s2hkbntel.net:5060 Contact: sip:3594410...@113.253.230.26 sip%3a3594410...@113.253.230.26 Call-ID: 34c9241622c72c7d26b13fdc22d95...@s2hkbntel.net CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.10 Date: Sun, 15 Aug 2010 13:47:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 241 v=0 o=root 2083113394 2083113394 IN IP4 113.253.230.26 s=Asterisk PBX 1.6.2.10 c=IN IP4 113.253.230.26 t=0 0 m=audio 11316 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- -- Called 13398560...@hkbn2b --- SIP read from UDP:203.80.89.139:5060 --- SIP/2.0 100 Trying t: sip:13398560...@s2hkbntel.net:5060 f: ck...@mobile sip:3594410...@s2hkbntel.netsip%3a3594410...@s2hkbntel.net ;tag=as1d554c43 i: 34c9241622c72c7d26b13fdc22d95...@s2hkbntel.net CSeq: 102 INVITE v: SIP/2.0/UDP 113.253.230.26:5060 ;received=113.253.230.70;rport;branch=z9hG4bK575022bd Server: MCS5x00_3.0 k: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec l: 0 - --- (9 headers 0 lines) --- --- SIP read from UDP:203.80.89.139:5060 --- SIP/2.0 487 Request Terminated t: sip:13398560...@s2hkbntel.net:5060;tag=1652716799 f: ck...@mobile sip:3594410...@s2hkbntel.netsip%3a3594410...@s2hkbntel.net ;tag=as1d554c43 i: 34c9241622c72c7d26b13fdc22d95...@s2hkbntel.net CSeq: 102 INVITE v: SIP/2.0/UDP 113.253.230.26:5060 ;received=113.253.230.70;rport;branch=z9hG4bK575022bd k: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec l: 0 - --- (8 headers 0 lines) --- Transmitting (NAT) to 203.80.89.139:5060: ACK sip:13398560...@s2hkbntel.net:5060 SIP/2.0 Via: SIP/2.0/UDP 113.253.230.26:5060;branch=z9hG4bK575022bd;rport Max-Forwards: 70 From: ck...@mobile sip:3594410...@s2hkbntel.netsip%3a3594410...@s2hkbntel.net ;tag=as1d554c43 To: sip:13398560...@s2hkbntel.net:5060;tag=1652716799 Contact: sip:3594410...@113.253.230.26 sip%3a3594410...@113.253.230.26 Call-ID: 34c9241622c72c7d26b13fdc22d95...@s2hkbntel.net CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.2.10 Content-Length: 0 --- Scheduling destruction of SIP dialog ' 34c9241622c72c7d26b13fdc22d95...@s2hkbntel.net' in 6400 ms (Method: INVITE) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [998560...@dlpn_dp1:2] Hangup(SIP/6100-005b, ) in new stack == Spawn extension (DLPN_DP1, 998560848, 2) exited non-zero on 'SIP/6100-005b' Really destroying SIP dialog '34c9241622c72c7d26b13fdc22d95...@s2hkbntel.net' Method: INVITE ns*CLI sip set debug off SIP Debugging Disabled -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Use of Storage Area Network with Asterisk
Are there any best practices for using a SAN with Asterisk? In the past we've kept config files local, but voicemail on a SAN. Aree there any issues with latency putting voice prompts, configs, etc. on a SAN? Anyone have some best practices to share? MD -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Use of Storage Area Network with Asterisk
On Sun, Aug 15, 2010 at 9:09 AM, Michelle Dupuis mdup...@ocg.ca wrote: Are there any best practices for using a SAN with Asterisk? In the past we've kept config files local, but voicemail on a SAN. Aree there any issues with latency putting voice prompts, configs, etc. on a SAN? Anyone have some best practices to share? We mount up a Netapp SAN for backup purposes. We rsync the live files (/etc, /var/spool/asterisk) to the SAN hourly for backup (losing an hour of voicemail wouldn't hurt us that much), but you could rsync at a different frequency. But all live files Asterisk uses, including voice prompts, are served out of the local file system on top of RAID-1 local disk. We did this to allow Asterisk to continue functioning in he midst of a SAN/network outage - backups will error out or hang, but Asterisk will keep going. We push out voice prompts and most config files via Puppet ( http://www.puppetlabs.com/) - with the Puppet repository being backed by an SVN repository so we have version control of all the changes we push out. We do this for other systems (such as web servers) to ensure all the systems end up with the same versions of files as each other. The only downside is they don't all get the changes at exactly the same time, but for something like voice prompts and configs I would think that won't matter (voicemail is a different beast). As for voicemail, if I was running redundant voicemail servers, I'd probably do things differently - put the voicemails on a SAN of some kind, perhaps even modifying Asterisk (with the voice mail left hook) to copy any new voicemail to the other box after it is left, if the other box is responsive. Then, I would write something that could merge two voicemail stores (message 1 on VM store 1 might not be the same as message 1 on VM store 2 - if not, copy it over as a new message, not overwriting the old one). My principle with this has been Don't make Asterisk depend on anything it doesn't absolutely have to depend upon. But I do think you could run prompts and configs off of a SAN - no problem there - but just that you would be building a dependency that would cause Asterisk to have issues if the SAN went offline or became unreachable. How reliable is your network/SAN? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Realtime Context
Hi, I'd like to be able to create contexts in real-time when I add new clients to my asterisk box. Currently, I have to create a blank context in extensions.conf and add:- switch = Realtime/@ Is there any way to avoid the step of creating the blank context and simply include all the entries from the database? Thanks Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fwd: 603 error
Hi, I have an interesting problem that the dial out via sip always generates 603 error The following is the sip debug Your help is appreciated. CK == Using SIP RTP CoS mark 5 -- Executing [998560...@dlpn_dp1:1] Dial(SIP/6100-005b, SIP/13398560...@hkbn2b) in new stack == Using SIP RTP CoS mark 5 Audio is at 113.253.230.26 port 11316 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 203.80.89.139:5060: INVITE sip:13398560...@s2hkbntel.net:5060 SIP/2.0 Via: SIP/2.0/UDP 113.253.230.26:5060;branch=z9hG4bK575022bd;rport Max-Forwards: 70 From: ck...@mobile sip:3594410...@s2hkbntel.netsip%3a3594410...@s2hkbntel.net ;tag=as1d554c43 To: sip:13398560...@s2hkbntel.net:5060 Contact: sip:3594410...@113.253.230.26 sip%3a3594410...@113.253.230.26 Call-ID: 34c9241622c72c7d26b13fdc22d95...@s2hkbntel.net CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.10 Date: Sun, 15 Aug 2010 13:47:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 241 v=0 o=root 2083113394 2083113394 IN IP4 113.253.230.26 s=Asterisk PBX 1.6.2.10 c=IN IP4 113.253.230.26 t=0 0 m=audio 11316 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- -- Called 13398560...@hkbn2b --- SIP read from UDP:203.80.89.139:5060 --- SIP/2.0 100 Trying t: sip:13398560...@s2hkbntel.net:5060 f: ck...@mobile sip:3594410...@s2hkbntel.netsip%3a3594410...@s2hkbntel.net ;tag=as1d554c43 i: 34c9241622c72c7d26b13fdc22d95...@s2hkbntel.net CSeq: 102 INVITE v: SIP/2.0/UDP 113.253.230.26:5060 ;received=113.253.230.70;rport;branch=z9hG4bK575022bd Server: MCS5x00_3.0 k: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec l: 0 - --- (9 headers 0 lines) --- --- SIP read from UDP:203.80.89.139:5060 --- SIP/2.0 487 Request Terminated t: sip:13398560...@s2hkbntel.net:5060;tag=1652716799 f: ck...@mobile sip:3594410...@s2hkbntel.netsip%3a3594410...@s2hkbntel.net ;tag=as1d554c43 i: 34c9241622c72c7d26b13fdc22d95...@s2hkbntel.net CSeq: 102 INVITE v: SIP/2.0/UDP 113.253.230.26:5060 ;received=113.253.230.70;rport;branch=z9hG4bK575022bd k: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec l: 0 - --- (8 headers 0 lines) --- Transmitting (NAT) to 203.80.89.139:5060: ACK sip:13398560...@s2hkbntel.net:5060 SIP/2.0 Via: SIP/2.0/UDP 113.253.230.26:5060;branch=z9hG4bK575022bd;rport Max-Forwards: 70 From: ck...@mobile sip:3594410...@s2hkbntel.netsip%3a3594410...@s2hkbntel.net ;tag=as1d554c43 To: sip:13398560...@s2hkbntel.net:5060;tag=1652716799 Contact: sip:3594410...@113.253.230.26 sip%3a3594410...@113.253.230.26 Call-ID: 34c9241622c72c7d26b13fdc22d95...@s2hkbntel.net CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.2.10 Content-Length: 0 --- Scheduling destruction of SIP dialog ' 34c9241622c72c7d26b13fdc22d95...@s2hkbntel.net' in 6400 ms (Method: INVITE) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [998560...@dlpn_dp1:2] Hangup(SIP/6100-005b, ) in new stack == Spawn extension (DLPN_DP1, 998560848, 2) exited non-zero on 'SIP/6100-005b' Really destroying SIP dialog '34c9241622c72c7d26b13fdc22d95...@s2hkbntel.net' Method: INVITE ns*CLI sip set debug off SIP Debugging Disabled -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
