[asterisk-users] ss7 installation and configuration
Hi, I'm unable to configure SS7 (surely my bad because it's my first try). I get this error: ERROR[15475] chan_dahdi.c: Unknown signalling method 'ss7' My system has: asterisk 1.4.31 zaptel 1.4.12.1 libpri 1.4.11.5 libss7 1.0.1 (installed from source) I can't upgrade this server to Dahdi and latest asterisk version... In any case, according to the libss7 README, it should work with my software versions. How can I make sure Asterisk is loading the SS7 library? According to libss7, I should place signalling=ss7 in /etc/asterisk/zapata.conf. Is that right? Do I need to recompile zaptel AFTER I install libss7? Thanks Vieri -- _ -- 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] Hint'ing with XMPP?
2011/12/7, Jamie A. Stapleton jstaple...@computer-business.com: Yes, we are using it. Most of the docs on the Internet are for 1.4. However, we now have it working with 1.8 (after some work). What do you imply by after some work ? Did you have to modify asterisk code ? asterisk config ? both ? ... -- _ -- 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] ss7 installation and configuration
On 12/07/2011 06:15 AM, Vieri wrote: I can't upgrade this server to Dahdi and latest asterisk version... In any case, according to the libss7 README, it should work with my software versions. What makes you think that? There is no support for SS7 in Asterisk 1.4. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- 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] SS7 + T1
I spoke with the Asterisk Pre-sales team and they said that SS7 support isn't technically supported, but it is there (e.g. talk to the OS community about this) so here's my question: I'm trying to interface an Asterisk Softswitch to a Nortel DMS100. If I get a dual-span card, can I run SS7 signaling over one span, and a T1 over the other span and have Asterisk link the two (e.g. caller-ID for the call on the 1st channel comes across the SS7... and so forth?)? -- _ -- 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] redirect a ringing phone
I am using AMI to call a phone and play a wave file. That works fine
to SIP/401.
Now I am trying to redirect that call that is ringing to another phone
(SIP/404).
When I do it the other phone rings but the first phone continues to ring
also.
Then when I answer on SIP/404, I get a ring not the wave file.
Action: Setvar
Channel: SIP/401-0004
Variable: SMVOICE_CALLAT
Value: SIP/404
DEBUG: Response: Success[CR ][LF ]Message: Variable Set[CR ][LF ][CR ][LF ]
Action: Redirect
Channel: SIP/401-0004
Exten: smvoice_dial_no_extension
Context: smvoice-transfers
Priority: 1
DEBUG: Response: Success[CR ][LF ]Message: Variable Set[CR ][LF ][CR ][LF ]
exten =
smvoice_dial_no_extension,1,Dial(${SMVOICE_CALLAT},${SMVOICE_DIAL_TIMEOUT},${SMVOICE_ONHOLD}tT)
What am I not issuing correctly to redirect the ringing phone to another
phone.
Thanks,
jerry
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Re: [asterisk-users] SS7 + T1
- Original Message - I spoke with the Asterisk Pre-sales team and they said that SS7 support isn't technically supported, but it is there (e.g. talk to the OS community about this) so here's my question: I'm trying to interface an Asterisk Softswitch to a Nortel DMS100. If I get a dual-span card, can I run SS7 signaling over one span, and a T1 over the other span and have Asterisk link the two (e.g. caller-ID for the call on the 1st channel comes across the SS7... and so forth?)? I'm personally not experienced with SS7. However, Sangoma[1] does provide support for SS7 with Asterisk. You may want to contact their sales/support personnel with your questions. --Tim [1] http://www.sangoma.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
[asterisk-users] Realtime Registration
[Dec 7 19:31:18] DEBUG[4217]: res_config_pgsql.c:821 pgsql_reconnect: Postgresql RealTime: Everything is fine. [Dec 7 19:31:18] DEBUG[4217]: res_config_pgsql.c:201 realtime_pgsql: Postgresql RealTime: Result=0x12e4edd0 Query: SELECT * FROM sippeers WHERE name = '105680' AND host = 'dynamic' [Dec 7 19:31:18] DEBUG[4217]: res_config_pgsql.c:209 realtime_pgsql: Postgresql RealTime: Found 1 rows. [Dec 7 19:31:18] ERROR[4217]: chan_sip.c:9940 register_verify: Peer '105680' is trying to register, but not configured as host=dynamic Any suggestions??? Asterisk 1.4.42 -- _ -- 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 Registration
[Dec 7 19:31:18] ERROR[4217]: chan_sip.c:9940 register_verify: Peer '105680' is trying to register, but not configured as host=dynamic Going on this, I'd say you probably tried to specify the host with a static IP address or a host name. If that's the case, you can't register, because that would be against the whole point of registering in the first place. You should probably post the DB entry for this peer to this thread to make things simpler... if it doesn't contain sensitive data. Of course, you can censor that out too. - Original Message - From: Andrew O. Zhukov gn...@telegroup.com.ua To: asterisk-users@lists.digium.com Sent: Wednesday, December 7, 2011 11:56:20 AM Subject: [asterisk-users] Realtime Registration [Dec 7 19:31:18] DEBUG[4217]: res_config_pgsql.c:821 pgsql_reconnect: Postgresql RealTime: Everything is fine. [Dec 7 19:31:18] DEBUG[4217]: res_config_pgsql.c:201 realtime_pgsql: Postgresql RealTime: Result=0x12e4edd0 Query: SELECT * FROM sippeers WHERE name = '105680' AND host = 'dynamic' [Dec 7 19:31:18] DEBUG[4217]: res_config_pgsql.c:209 realtime_pgsql: Postgresql RealTime: Found 1 rows. [Dec 7 19:31:18] ERROR[4217]: chan_sip.c:9940 register_verify: Peer '105680' is trying to register, but not configured as host=dynamic Any suggestions??? Asterisk 1.4.42 -- _ -- 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 -- _ -- 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] google voice calling dial plan question.
On Tue, Dec 6, 2011 at 4:05 PM, white hat whitehat...@gmail.com wrote: Would you be willing to post sanitized versions of your jabber.conf, gtalk.conf and details regarding the context you're using and how your inbound route is configured in your dial plan? Are you using STUN? Is Asterisk behind a NAT device or on a public IP? Yes, to both of the last questions. I am using STUN and my asterisk(s) are behind a NAT device (a Netgear WND3700). My jabber.conf looks like: [general] autoregister=yes debug=yes autoprune=no auth_policy=accept [asterisk] type=client serverhost=talk.google.com ; username=xxx...@gmail.com/Talk username=xx...@gmail.com/asterisk secret=XX priority=1 port=5222 usetls=yes usesasl=yes buddy=xxx...@gmail.com status=available statusmessage=I am an Asterisk Server timeout=100 context=gtalk_incoming and, gtalk.conf looks like this: [general] context=LocalSets ; Context to dump call into bindaddr=0.0.0.0; Address to bind to allowguests=yes ; Allow calls from people not in list of peers [guest] ; special account for options on guest account disallow=all allow=ulaw context=gtalk_incoming [XX] username=xxx...@gmail.com disallow=all allow=ulaw context=gtalk_incoming connection=asterisk And, I think that just dumps incoming calls into the context that I posted previously. HTH, dwa -- + dai...@pervasivetelcom.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 Registration
No secrets :) SELECT * FROM sippeers WHERE name = '105680' AND host = 'dynamic' name|type|username|secret|fromuser|fromdomain|nat|context|canreinvite|disallow|allow|host|insecure|port|ipaddr|outboundproxy 105680|peer|testbutton2|XXX||button.ipshka.com:5060|no|button|no|all|speex;ulaw;alaw;g729;ilbc;g726;g726aal2;slin;lpc10;adpcm;g723|dynamic|port,invite|5060 ||ipshka.com On 12/07/2011 08:04 PM, Jonathan Rose wrote: [Dec 7 19:31:18] ERROR[4217]: chan_sip.c:9940 register_verify: Peer '105680' is trying to register, but not configured as host=dynamic Going on this, I'd say you probably tried to specify the host with a static IP address or a host name. If that's the case, you can't register, because that would be against the whole point of registering in the first place. You should probably post the DB entry for this peer to this thread to make things simpler... if it doesn't contain sensitive data. Of course, you can censor that out too. - Original Message - From: Andrew O. Zhukovgn...@telegroup.com.ua To: asterisk-users@lists.digium.com Sent: Wednesday, December 7, 2011 11:56:20 AM Subject: [asterisk-users] Realtime Registration [Dec 7 19:31:18] DEBUG[4217]: res_config_pgsql.c:821 pgsql_reconnect: Postgresql RealTime: Everything is fine. [Dec 7 19:31:18] DEBUG[4217]: res_config_pgsql.c:201 realtime_pgsql: Postgresql RealTime: Result=0x12e4edd0 Query: SELECT * FROM sippeers WHERE name = '105680' AND host = 'dynamic' [Dec 7 19:31:18] DEBUG[4217]: res_config_pgsql.c:209 realtime_pgsql: Postgresql RealTime: Found 1 rows. [Dec 7 19:31:18] ERROR[4217]: chan_sip.c:9940 register_verify: Peer '105680' is trying to register, but not configured as host=dynamic Any suggestions??? Asterisk 1.4.42 -- _ -- 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 -- _ -- 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 -- _ -- 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] ChanSpy() and Spygroup
I am running an Asterisk 1.4.34 installation. I am trying to separate several SIP phones into two separate spygroups. These phones are making external calls as opposed to receiving incoming calls. Is there a place to assign a phone to a Spygroup other than when the call is initiated. I am trying to avoid having to make separate external call behaviours for every single extension in the dialplan that will be monitored. Thanks, Jeremy Jeremy Hellstrom Specialist, IT/Systems I 604.664.2472 I 604.664.2400 (245) | #1550-1090 W. Georgia St. Vancouver BC V6E 3V7 I www.synovate.com http://www.synovate.com/ State-of-the-art focus group facility, now with FocusVision! For recruiting, hosting and facilitating, visit www.vancouverfocusgroups.com http://www.vancouverfocusgroups.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] ChanSpy() and Spygroup
Just a thought - make your normal phone use context default and your others use context spyonme From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jeremy.hellst...@synovate.com Sent: Wednesday, December 07, 2011 2:11 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] ChanSpy() and Spygroup I am running an Asterisk 1.4.34 installation. I am trying to separate several SIP phones into two separate spygroups. These phones are making external calls as opposed to receiving incoming calls. Is there a place to assign a phone to a Spygroup other than when the call is initiated. I am trying to avoid having to make separate external call behaviours for every single extension in the dialplan that will be monitored. Thanks, Jeremy Jeremy Hellstrom Specialist, IT/Systems I 604.664.2472 I 604.664.2400 (245) | #1550-1090 W. Georgia St. Vancouver BC V6E 3V7 I www.synovate.com http://www.synovate.com/ State-of-the-art focus group facility, now with FocusVision! For recruiting, hosting and facilitating, visit www.vancouverfocusgroups.com http://www.vancouverfocusgroups.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
[asterisk-users] Help! Logs filling up with errors!
I am running Asterisk 1.6.2.20 with dahdi 2.5.0.2, Oslec from kernel sources. Hardware is 2 X100P Wildcards. Everything seems to be working OK but my logs are filling up with this message: Dec 7 14:25:06 servername kernel: FXO PCI Master abort The messages just pour in constantly until the hard drive is full. It's eaten 50+ gigs 4 times already today. OS is Centos 6.0, with custom kernel is 2.6.39.4.smp.x86_64. The motherboard is an Asus M4A78LT-M LE running a dual-core AMD CPU and 4gigs of ram. Does anyone know what might be causing this? Thanks, Brent -- _ -- 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] Help! Logs filling up with errors!
Check this post - it sounds like exactly what is happening to you. http://osdir.com/ml/debian.packages.voip.devel/2009-01/msg00046.html -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brent Davidson Sent: Wednesday, December 07, 2011 2:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Help! Logs filling up with errors! I am running Asterisk 1.6.2.20 with dahdi 2.5.0.2, Oslec from kernel sources. Hardware is 2 X100P Wildcards. Everything seems to be working OK but my logs are filling up with this message: Dec 7 14:25:06 servername kernel: FXO PCI Master abort The messages just pour in constantly until the hard drive is full. It's eaten 50+ gigs 4 times already today. OS is Centos 6.0, with custom kernel is 2.6.39.4.smp.x86_64. The motherboard is an Asus M4A78LT-M LE running a dual-core AMD CPU and 4gigs of ram. Does anyone know what might be causing this? Thanks, Brent -- _ -- 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 -- _ -- 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] Help! Logs filling up with errors!
On 12/7/2011 2:35 PM, Danny Nicholas wrote: Check this post - it sounds like exactly what is happening to you. http://osdir.com/ml/debian.packages.voip.devel/2009-01/msg00046.html -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brent Davidson Sent: Wednesday, December 07, 2011 2:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Help! Logs filling up with errors! I am running Asterisk 1.6.2.20 with dahdi 2.5.0.2, Oslec from kernel sources. Hardware is 2 X100P Wildcards. Everything seems to be working OK but my logs are filling up with this message: Dec 7 14:25:06 servername kernel: FXO PCI Master abort The messages just pour in constantly until the hard drive is full. It's eaten 50+ gigs 4 times already today. OS is Centos 6.0, with custom kernel is 2.6.39.4.smp.x86_64. The motherboard is an Asus M4A78LT-M LE running a dual-core AMD CPU and 4gigs of ram. Does anyone know what might be causing this? Thanks, Brent -- Yes, that appears to be what is happening to me, but I can't seem to find a solution. -- _ -- 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] Help! Logs filling up with errors!
On Wed, Dec 07, 2011 at 02:46:51PM -0600, Brent Davidson wrote: On 12/7/2011 2:35 PM, Danny Nicholas wrote: Check this post - it sounds like exactly what is happening to you. http://osdir.com/ml/debian.packages.voip.devel/2009-01/msg00046.html Yes, that appears to be what is happening to me, but I can't seem to find a solution. There are compatibility problems with the interface chip used on the X100P and certains hosts. ZAP-113 [1] references someone who was able to solve a similar problem by ensuring that the cards aren't sharing interrupts. [1] https://issues.asterisk.org/jira/browse/ZAP-113 If everything *really* is just working fine for you (which I would doubt) you could always just comment out the line from drivers/dahdi/wcfxo.c (line 465) and at least then your logs won't fill up. But I think this would just end up masking another problem. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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] ISDN PRI configuration
Hi, A telco has recently installed a new line in our building and I need to connect it to my Asterisk server with a Digium PRI card. It's not the first time I set up and configure a PRI link but I'm failing to make this one work. The only information I got from the telco is: Line Coding [HDB3] Framing [CRC4] Encapsultation [hdlc Isdn switch-type primary-[net5] Is crc4 actually a framing parameter as stated by the telco, or is it just an optional line coding parameter? I searched the web and not knowing exactly which parameters to use, I tried the following zaptel/dahdi config: # TE120P (PRI): span=1,1,0,ccs,hdb3,crc4 # as E1 bchan=1-15 dchan=16 bchan=17-31 switchtype = euroisdn signalling = pri_cpe However, the link doesn't work and I get this: *CLI show status: Description Alarms IRQbpviol CRC4 Wildcard TE120P Card 0 RED1 0 0 # cat /proc/zaptel/1 Span 1: WCT1/0 Wildcard TE120P Card 0 (MASTER) HDB3/CCS/CRC4 RED IRQ misses: 1 1 WCT1/0/1 Clear (In use) RED 2 WCT1/0/2 Clear (In use) RED 3 WCT1/0/3 Clear (In use) RED 4 WCT1/0/4 Clear (In use) RED 5 WCT1/0/5 Clear (In use) RED 6 WCT1/0/6 Clear (In use) RED 7 WCT1/0/7 Clear (In use) RED 8 WCT1/0/8 Clear (In use) RED 9 WCT1/0/9 Clear (In use) RED 10 WCT1/0/10 Clear (In use) RED 11 WCT1/0/11 Clear (In use) RED 12 WCT1/0/12 Clear (In use) RED 13 WCT1/0/13 Clear (In use) RED 14 WCT1/0/14 Clear (In use) RED 15 WCT1/0/15 Clear (In use) RED 16 WCT1/0/16 HDLCFCS (In use) RED 17 WCT1/0/17 Clear (In use) RED 18 WCT1/0/18 Clear (In use) RED 19 WCT1/0/19 Clear (In use) RED 20 WCT1/0/20 Clear (In use) RED 21 WCT1/0/21 Clear (In use) RED 22 WCT1/0/22 Clear (In use) RED 23 WCT1/0/23 Clear (In use) RED 24 WCT1/0/24 Clear (In use) RED 25 WCT1/0/25 Clear (In use) RED 26 WCT1/0/26 Clear (In use) RED 27 WCT1/0/27 Clear (In use) RED 28 WCT1/0/28 Clear (In use) RED 29 WCT1/0/29 Clear (In use) RED 30 WCT1/0/30 Clear (In use) RED 31 WCT1/0/31 Clear (In use) RED Placing a call through the Zap/Dahdi trunk in Asterisk doesn't work and I get the following message in the log: chan_dahdi.c: No D-channels available! Using Primary channel 16 as D-channel anyway! logger.c: -- Attempting call on Zap/g1/999xx for 999xx@custom-TESTCALL:1 (Retry 1) channel.c: Unable to request channel Zap/g1/999xx pbx_spool.c: Call failed to go through, reason (8) Congestion (circuits busy) chan_dahdi.c: No D-channels available! Using Primary channel 16 as D-channel anyway! Am I missing some information here? I'm *supposing* it should be E1 (and that I can use 16 as dchan), euroisdn (not national), but my telco states hdlc Isdn switch-type primary-[net5] and I don't know how to translate it to zaptel/dahdi... Also, my telco hasn't mentioned anything about ccs but I tried it anyway because I wouldn't know what else to use. I also tried signalling = pri_net but still got the same RED alerts. Any suggestions? Thanks Vieri -- _ -- 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] ISDN PRI configuration
On Wed, 7 Dec 2011, Vieri wrote: A telco has recently installed a new line in our building and I need to connect it to my Asterisk server with a Digium PRI card. It's not the first time I set up and configure a PRI link but I'm failing to make this one work. chan_dahdi.c: No D-channels available! Using Primary channel 16 as D-channel anyway! We usually get D channels on the first channel of the first T1 in an NFAS group and the last channel of the last t1. However, telcos don't always get the order right. I've spent hours trying configurations and varying the D channel. Sometimes it's just that they number things in a different order than we were expecting. Sometimes, it almost appears that they use a dartboard :) -- 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 -- 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] ISDN PRI configuration
On 12/07/2011 04:15 PM, Steve Edwards wrote: On Wed, 7 Dec 2011, Vieri wrote: A telco has recently installed a new line in our building and I need to connect it to my Asterisk server with a Digium PRI card. It's not the first time I set up and configure a PRI link but I'm failing to make this one work. chan_dahdi.c: No D-channels available! Using Primary channel 16 as D-channel anyway! We usually get D channels on the first channel of the first T1 in an NFAS group and the last channel of the last t1. However, telcos don't always get the order right. I've spent hours trying configurations and varying the D channel. Sometimes it's just that they number things in a different order than we were expecting. Sometimes, it almost appears that they use a dartboard :) Vieri: You aren't even far enough along to worry about D-channel assignments or anything like that. Your span is in RED alarm; that means it can't see the far end at all. Until you get that cured (layer 1 - physical layer) nothing above it is going to work. Since they mentioned HDB3 and CRC4, you most definitely have an E1 span, and you will need to specify 'CCS' as well because you are using ISDN signaling. If the line coding/framing settings are wrong that *could* result in a RED alarm, but doesn't always. So, you need to start by getting the span to come out of RED alarm (to go 'green'). This could be a cabling problem, a hardware problem, or it could something as simple as the fact that the telco hasn't actually 'turned up' the span yet, because they don't usually do that until you have your equipment plugged in and you call them to tell them that you are ready for the span to be turned up. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- 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] ISDN PRI configuration
--- On Wed, 12/7/11, Steve Edwards asterisk@sedwards.com wrote: A telco has recently installed a new line in our building and I need to connect it to my Asterisk server with a Digium PRI card. It's not the first time I set up and configure a PRI link but I'm failing to make this one work. chan_dahdi.c: No D-channels available! Using Primary channel 16 as D-channel anyway! We usually get D channels on the first channel of the first T1 in an NFAS group and the last channel of the last t1. However, telcos don't always get the order right. I've spent hours trying configurations and varying the D channel. Sometimes it's just that they number things in a different order than we were expecting. Sometimes, it almost appears that they use a dartboard :) As far as I know, E1 usually use 16 as D channel. Anyway, I tried as you suggested and set 1 as the D channel and 2-31 as B channels. In the asterisk log I got these messages: chan_dahdi.c: Channel 16 is reserved for D-channel. chan_dahdi.c: Unable to register channel '2-31' So doesn't this actually tell me that I should keep using 16 as the D channel? (so chan_dahdi actually knows about it on its own, I guess) It's funny though that chan_dahdi tells me I have to use channel 16 as D channel whenever I try to use another one, but when I do use 16, it says that there are no D channels available. Confusing. Thanks anyway for the reply. Vieri -- _ -- 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] ISDN PRI configuration
--- On Wed, 12/7/11, Kevin P. Fleming kpflem...@digium.com wrote: Vieri: You aren't even far enough along to worry about D-channel assignments or anything like that. Your span is in RED alarm; that means it can't see the far end at all. Until you get that cured (layer 1 - physical layer) nothing above it is going to work. Since they mentioned HDB3 and CRC4, you most definitely have an E1 span, and you will need to specify 'CCS' as well because you are using ISDN signaling. If the line coding/framing settings are wrong that *could* result in a RED alarm, but doesn't always. So, you need to start by getting the span to come out of RED alarm (to go 'green'). This could be a cabling problem, a hardware problem, or it could something as simple as the fact that the telco hasn't actually 'turned up' the span yet, because they don't usually do that until you have your equipment plugged in and you call them to tell them that you are ready for the span to be turned up. They should have turned it up, or at least that's what one of the tech guys told me. But I guess I'll have to check with them again. The cable should be ok (standard ethernet cable) but I didn't actually install it myself (I'm in a remote location) so I'll have to check that too. Big thanks for the explanation! Vieri -- _ -- 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] ISDN PRI configuration
On 12/07/2011 04:51 PM, Vieri wrote: --- On Wed, 12/7/11, Kevin P. Flemingkpflem...@digium.com wrote: Vieri: You aren't even far enough along to worry about D-channel assignments or anything like that. Your span is in RED alarm; that means it can't see the far end at all. Until you get that cured (layer 1 - physical layer) nothing above it is going to work. Since they mentioned HDB3 and CRC4, you most definitely have an E1 span, and you will need to specify 'CCS' as well because you are using ISDN signaling. If the line coding/framing settings are wrong that *could* result in a RED alarm, but doesn't always. So, you need to start by getting the span to come out of RED alarm (to go 'green'). This could be a cabling problem, a hardware problem, or it could something as simple as the fact that the telco hasn't actually 'turned up' the span yet, because they don't usually do that until you have your equipment plugged in and you call them to tell them that you are ready for the span to be turned up. They should have turned it up, or at least that's what one of the tech guys told me. But I guess I'll have to check with them again. The cable should be ok (standard ethernet cable) but I didn't actually install it myself (I'm in a remote location) so I'll have to check that too. Standard Ethernet cables do not always work for T-1/E-1 spans. They do work a rather large percentage of the time, but not always. Distance between the NIU and the T-1/E-1 card can be a factor, among other things. Many Digium products include span loopback devices, that you can plug a cable into and generate a hard loopback towards the card. If there is one of those on-site, have someone unplug the cable from the NIU and plug it into the loopback device instead; if the span goes green, then at least your cabling/wiring are OK. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- 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] ISDN PRI configuration
and maybe more but right now I don't recall any loopback device although I won't be sure until I go to the site. Can a loopback device be bought seperately? Sure, we use the below device all the time: http://www.amazon.com/Cables-Unlimited-TST-LOOP-003-Smartronix-Superlooper/dp/B000V6Y7IY What kind of cable should be used instead of an ethernet cable (I think they used a 5m long cat5 T-568B Straight-Through Ethernet Cable)? Thanks, Vieri -- _ -- 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 -- Technical Support http://www.telesip.net -- _ -- 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] Confrence call is not make
Hi, I am making confrence application. In sip.conf [phone1] type=friend host=dynamic Takes an alphanumeric string. context= employees [phone2] type=friend host=dynamic context= employees [phone3] type=friend host=dynamic context= employees In extension.conf [employees] exten = 101,1,Dial(SIP/phone1,20,tT) exten = 102,1,Dial(SIP/phone2,20,tT) exten = 103,1,Dial(SIP/phone3,20,tT) exten = 777,1,MeetMe(777) In meetme.conf [rooms] conf = 777 when i call 777 from phone1 ,its shows 603 declined. I check in CLI [Dec 6 17:46:58] WARNING[16264]: pbx.c:4088 pbx_extension_helper: No application 'MeetMe' for extension (employees, 777, 1) == Spawn extension (employees, 777, 1) exited non-zero on 'SIP/phone1-' Plz tell me , where i am wrong in configuration. Thanks -- -- _ -- 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] Confrence call is not make
On Thu, 8 Dec 2011, Durgesh Mishra wrote: I am making confrence application. [Dec 6 17:46:58] WARNING[16264]: pbx.c:4088 pbx_extension_helper: No application 'MeetMe' for extension (employees, 777, 1) You don't have app_meetme.so loaded. What happens if you enter 'module load app_meetme.so' at Asterisk's CLI? -- 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 -- 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] Confrence call is not make
I check in CLI [Dec 6 17:46:58] WARNING[16264]: pbx.c:4088 pbx_extension_helper: No application 'MeetMe' for extension (employees, 777, 1) == Spawn extension (employees, 777, 1) exited non-zero on 'SIP/phone1-' Plz tell me , where i am wrong in configuration. Chances are you didn't install DAHDI before building Asterisk, so you never built the MeetMe application. Go download install DAHDI, then rerun ./configure menuselect for Asterisk and make sure the MeetMe app is available. -- _ -- 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] Reg:Asterisk can be installed in Oracle Enterprise Linux 6 OS
Hello List, Can anyone please confirm whether Asterisk can be installed in Oracle Enterprise Linux6 OS. Thanks, Rekha -- _ -- 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] Confrence call is not make
Hi This is due to module app_meetme.so is not loaded. Execute below command in asterisk cli and check the cli logger. module load app_meetme.so If you are installed asterisk in a linux system without any analog interface this meetme application will not work. You have use application Conference instead of MeetMe. Thanks Nikhil On 12/08/2011 11:12 AM, Durgesh Mishra wrote: Hi, I am making confrence application. In sip.conf [phone1] type=friend host=dynamic Takes an alphanumeric string. context= employees [phone2] type=friend host=dynamic context= employees [phone3] type=friend host=dynamic context= employees In extension.conf [employees] exten = 101,1,Dial(SIP/phone1,20,tT) exten = 102,1,Dial(SIP/phone2,20,tT) exten = 103,1,Dial(SIP/phone3,20,tT) exten = 777,1,MeetMe(777) In meetme.conf [rooms] conf = 777 when i call 777 from phone1 ,its shows 603 declined. I check in CLI [Dec 6 17:46:58] WARNING[16264]: pbx.c:4088 pbx_extension_helper: No application 'MeetMe' for extension (employees, 777, 1) == Spawn extension (employees, 777, 1) exited non-zero on 'SIP/phone1-' Plz tell me , where i am wrong in configuration. Thanks -- -- _ -- 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 -- _ -- 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] How to make app_meetme enable
In make menuselect =application=XXX app_meetme . I am doing confrence call using sip softphone. I checked It Depends on: dahdi(E) . How I can do app_meetme enable? Thanks -- _ -- 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] Talk detection in meetme
??? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eyal Sent: Tuesday, December 06, 2011 11:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Talk detection in meetme Hi, I create Chat room with MEETME and now I have a problem. I want that the host of the room could identify the participants in the room by their speech, so that if a participant uses language the host could kick him from the room. Is there a way to do it? thanks. Eyal Mahalal -- _ -- 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
