Re: [Asterisk-Users] Asterisk LDAP Authentication Problem
did you notice the two dots in the IP address of ldaphost ? Marco. Chandan Mishra wrote: Hi I want to authenticate the asterisk users from the LDAP directory server not from the sip.conf. I tried to use the astirectory-1.2 http://www..asterisk-ev.org/download/astirectory-1.2-0.3.tgz . But i am not able to configure it properly. If somebody used it then please help. In the res_ldap.conf file i made the following entries. I am using my normal username and password to connect my asterisk server to the LDAP server of my organization. Is some administrator login is requried to connect ? [general] ldapuser=cn=chandan.mishra,dc=synapse,dc=com ldaphost=ldap://192.168.0..16 ldap://192.168.0.16 ldappass=chandan123 ldapbasedn=dc=synapse,dc=com After this the asterisk is not able to connect to the ldap database. And hence asterisk is not able to start. Its giving me following errors: == Parsing '/etc/asterisk/res_ldap.conf': Found Jan 17 23:38:09 WARNING[11207]: res_config_ldap.c:615 parse_config: LDAP RealTime: No database host found, using localhost via socket. Jan 17 23:38:09 WARNING[11207]: res_config_ldap.c:630 parse_config: LDAP RealTime Host: ldap://localhost Jan 17 23:38:09 WARNING[11207]: res_config_ldap.c:631 parse_config: LDAP RealTime User: cn= chandan.mishra,dc=synapse,dc=com Jan 17 23:38:09 WARNING[11207]: res_config_ldap.c:632 parse_config: LDAP RealTime Base DN: dc=synapse,dc=com Jan 17 23:38:09 ERROR[11207]: res_config_ldap.c:708 ldap_reconnect: LDAP failed to bind (host= ldap://localhost, user=cn=chandan.mishra,dc=synapse,dc=com Can't contact LDAP server 0)! Jan 17 23:38:09 ERROR[11207]: res_config_ldap.c:708 ldap_reconnect: LDAP failed to bind (host= ldap://localhost, user=cn=chandan.mishra,dc=synapse,dc=com Can't contact LDAP server 1)! Jan 17 23:38:09 ERROR[11207]: res_config_ldap.c:708 ldap_reconnect: LDAP failed to bind (host=ldap://localhost ldap://localhost, user=cn=chandan.mishra,dc=synapse,dc=com Can't contact LDAP server 2)! Jan 17 23:38:09 WARNING[11207]: res_config_ldap.c:539 load_module: LDAP RealTime: Couldn't establish connection. Check debug. Thanks Chandan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CallProgress breaks DTMF
Hi, I enabled Callprogress in the zapata.conf , so in the CDR it will log other things other then answered (Busy, no answer etc), but, this seems to break my Polycom's DTMF, i configured RFC2833 for the dtmf in the sip.conf, and when callprogress is enabled, the dtmf doesnt reach the other end, Any idea/solution ? Thanks. Marco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CallProgress breaks DTMF - RFC2833
Hi I enabled Callprogress in the zapata.conf , so in the CDR it will log other things other then answered (Busy, no answer etc), but, this seems to break my Polycom's DTMF, i configured RFC2833 for the dtmf in the sip.conf, and when callprogress is enabled, the dtmf doesnt reach the other end, Any idea/solution ? Thanks. Marco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hangup detection - TDM400P
Hi, I have a long delay when detecting hangups on the TDM400P card, with 4 FXO ports, When an incoming call dial's in, when hanging up, the asterisk will detect the hangup only after 10 seconds, i searched around, and found many similar problems, but no solution, i tried some options in zapate.conf , but nothing helped, any solution ? the lines are coming from SBC in San Fransisco, i asked them if i have disconnect supervision, and they said i do have it. Marco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CallerID Length
Hi, I have a problem with the Caller ID string, seems like asterisk will display only 10 digits of the caller id. If the string is longer then 10 digits, asterisk will sometimes strip the first digit, and sometimes the last digits, in order to show a 10-digit callerid, Is this configurable ? i would like to get the caller id of international callers , with all digits. Any solution ? Thanks. Marco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hangup detection - TDM400P
Yes, didnt change anything Marco. Angelito Manansala wrote: hmmm di you try this ;hanguponpolarityswitch=yes Cheerz! On 11/17/05, Marco Supino [EMAIL PROTECTED] wrote: Hi, I have a long delay when detecting hangups on the TDM400P card, with 4 FXO ports, When an incoming call dial's in, when hanging up, the asterisk will detect the hangup only after 10 seconds, i searched around, and found many similar problems, but no solution, i tried some options in zapate.conf , but nothing helped, any solution ? the lines are coming from SBC in San Fransisco, i asked them if i have disconnect supervision, and they said i do have it. Marco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PRI pass-through
Hi, I want to build a PRI pass-through with a Cisco 2600, with two VWIC E1 cards, is this possible ? and do i need any other modules except for the E1 modules ? What i want to do is connect the asterisk to the PRI through the Cisco router, and let my legacy PBX utilize some of the PRI channels while testing Asterisk, Anyone with experience, sample configs or idea, please contribute. Thanks. Marco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Detect registered peers
Hi, Is there a way to detect (in the dialplan) if a SIP peer is registered with the server ? I am using macros to dial to extension, becuase i dont want to define each extension in the dialplan, and, for example, my numbers are 8xx , i want to know if a peer exists/registered before ringing the line, i need something like Voicemailexists , but for SIP peers. any solution ? Thanks. Marco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fxotune fails with valid TDM/FXO card
Hi, I am using Asterisk 1.0.9 with the 1.2.0 zaptel, just for the fxotune utility, which solved my echo problems , my zttest results are low, but no echo on ZAP lines... Marco. Chris Miller wrote: Mojo with Horan Company, LLC wrote: The recent suggestion on the list was to not use 1.0.9 zaptel You mean the driver, or the version of fxotune? fxotune has been removed from the prior versions of the zaptel driver, it's only included in 1.2 now. As for the driver, is anyone using the 1.2 zaptel driver with Asterisk 1.0.9? The way the downloads are grouped together on the Asterisk web page, I was led to believe they shouldn't be mixed. Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] App_directory + Festival
Hi, As anyone tried integrating App_Directory with any Text2Speech mechanism like festival ? Marco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice Outages?
Yes, i am having timeouts on registering to the LAX sip server of broadvoice. Marco. Nate Kapi wrote: I've been having a lot of problems with Broadvoice lately. Anyone else been without service for extended periods of time this week? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zttest - 100% ?
Hi, I would like to know what type of configuration could get me closer to 100% hits in zttest, when testing a TDM400P with 4 FXO ports, I am currently running kernel 2.4.31, on a IBM Xseries 306, with 3gh CPU, HT is disabled, PCI latency was changed, i still cant get more then 99.975% in the zttest testings, Thanks for any info. Marco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zttest - 100% ?
Hi, My TDM is on its own IRQ, and the x306 has only one full-size PCI slot.. so no playing with it, what results do you get from zttest ? what IRQ is the card on ? Marco. Damian Funnell wrote: Have you checked that the TDM400P isn't sharing an IRQ with anything else? Don't trust /proc/interrupts - run lspci -v to confirm this. We have * running on an x206 and found that the only way to stop the TDP400P sharing an IRQ with other devices was to juggle cards between slots. Hope this helps! Damian. FFF Managed Technology Ltd 60 Cook St P.O. 6368 Wellesley St Auckland t +64 9 356 2911 f +64 9 358 9070 m +64 21 415 297 w www.fff.co.nz Marco Supino wrote: Hi, I would like to know what type of configuration could get me closer to 100% hits in zttest, when testing a TDM400P with 4 FXO ports, I am currently running kernel 2.4.31, on a IBM Xseries 306, with 3gh CPU, HT is disabled, PCI latency was changed, i still cant get more then 99.975% in the zttest testings, Thanks for any info. Marco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IBM x306 - some progress
Hi, I asked yesterday about a problem with x306 and IRQ sharing, didnt get much info, now, i was playing with lspci, and see something strange, lspci -v shows me the TDM400P card is on IRQ 7, and the SCSI card is also on IRQ 7, lspci -bv (from the man - b - shows bus-centric view, as seen by the BUS and not by the kernel) shows me the TDM400P is on IRQ 5, why does the kernel puts it on IRQ 7 ? any insights much appriciated. Marco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IBM x306
Hi, This is a little off-topic,but if someone has any info, it could help me a LOT!, I am building an asterisk pbx (1.0.9) on an IBM x306 SCSI machine,my problem is that the BIOS assigns the same IRQ to the SCSI controller, and the TDM400P, i have tried several options of making the bios change the IRQ, but it will always move them together, anyone with some info about my options ? Thanks, Marco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IBM x306
Only one PCI slot can hold the full size card like the TDM400P , the other slot has a smaller opening on the case. Marco. Alexander Lopez wrote: Can you try a different slot on the PCI bus?? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marco Supino Sent: Saturday, September 24, 2005 8:51 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] IBM x306 Hi, This is a little off-topic,but if someone has any info, it could help me a LOT!, I am building an asterisk pbx (1.0.9) on an IBM x306 SCSI machine,my problem is that the BIOS assigns the same IRQ to the SCSI controller, and the TDM400P, i have tried several options of making the bios change the IRQ, but it will always move them together, anyone with some info about my options ? Thanks, Marco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IBM x306
Hi, I tried setpci INTERRUPT_LEVEL (or something similar, cant remmeber now), and also setpci seems like it changed the IRQ, lspci -v still shows the old IRQ Marco. Stefan de Konink wrote: On Sun, 25 Sep 2005, Marco Supino wrote: I am building an asterisk pbx (1.0.9) on an IBM x306 SCSI machine,my problem is that the BIOS assigns the same IRQ to the SCSI controller, and the TDM400P, i have tried several options of making the bios change the IRQ, but it will always move them together, anyone with some info about my options ? Linux usually don't care about Bios settings, you could try kernel cmdline parameters. Acpi and IRQ are google terms for it. Stefan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoiceXML
Hi, Anyone has a working example of VoiceXML with asterisk ? i was looking around voip-info and the internet, and couldnt find more then proof of concept documents. Also, does anyone knows how FWD does their VoiceXML (411) service ? Thanks for any info Marco. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chan_modem_*
Hi, I was looking for solutions for simple FXO cards, and came across the two modem channels in the asterisk channels/ dir, i assume they are there becuase someone made these two types of modems work as FXO (or are they there for other purpose ?), does anyone have any info on these channels ? anyone has them working with any type of modem ? (aopen or bestdata). Marco. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Need info : lspci
Hi, I need some info from people with the x100p card (digium or clone), please send me the output of lspci and lspci -n from your linux machine, i am tring to find out something on my * server. Thanks. Marco. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] possible bug in chan_capi concerning context handling
Do you have an 's' extention in the default context ?
Marco.
Dimitris Kounalakis wrote:
Hello,
I am trying to configure asterisk 1.0.7pre to get incoming calls from an
ISDN line using an AVM fritz PCI 2.0 with Chan_capi 0.3.5. My problem is
that the context is not recognised in the /etc/asterisk/capi.conf
I have in /etc/asterisk/capi.conf 's section [interfaces] the
following directive
context=isdn
and the following directive in /etc/asterisk/extensions.conf in the
context [isdn]
[isdn]
exten = s,1,Dial(SIP/${DNID:4},60,tr)
Here follows the debug info I get when an incoming call starts:
== CONNECT_IND
(PLCI=0x101,DID=2810111694,CID=2810111694,CIP=0x1,CONTROLLER=0x1)
-- creating pipe for PLCI=0x101 msn = 2810111694
sent ALERT_REQ PLCI = 0x101
== Starting CAPI[contr1/2810111694]/3 at ,2810111694,1 failed so
falling back to exten 's'
== Starting CAPI[contr1/2810111694]/3 at ,s,1 still failed so falling
back to context 'default'
Mar 13 11:52:41 WARNING[10744]: pbx.c:1893 ast_pbx_run: Channel
'CAPI[contr1/2810111694]/3' sent into invalid extension 's' in context
'default', but no invalid handler
-- CAPI Hangingup
-
When I move the exten = s,1,Dial(${DNID:4},60,tr) in the context
[default] of the /etc/asterisk/extensions.conf, I get the following
debug info and the sip phone rings ok:
--
== CONNECT_IND
(PLCI=0x101,DID=2810111694,CID=2810111694,CIP=0x1,CONTROLLER=0x1)
-- creating pipe for PLCI=0x101 msn = 2810111694
sent ALERT_REQ PLCI = 0x101
== Starting CAPI[contr1/2810111694]/4 at ,2810111694,1 failed so
falling back to exten 's'
== Starting CAPI[contr1/2810111694]/4 at ,s,1 still failed so falling
back to context 'default'
-- Executing Dial(CAPI[contr1/2810111694]/4, SIP/111694|60|tr) in
new stack
-- Called 111694
--
Is this a bug? It does not handle the context, so, it can not find what
to do, it works only with the default context.
Thank you in advance,
Dimitris
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[Asterisk-Users] SIP registration problem
Hi, I am adding phones to my asterisk setup, until now i worked with some softphones, with no problem, I got some Grandstream BT100 phones, and see something strange in the log, the on the phone's screen, This is from the log : Found peer '122' Looking for 122 in default Transmitting (no NAT): SIP/2.0 404 Not Found This happends when the action is SUBSCRIBE , Now, this is a SIP client, defined in the sip.conf, as [122] context=default ... and also the exten is in the default context in the extension conf file, Right after the the peer seems to be registered, and the phone seems to work, but from time to time, i see 404 on the phone's display, and need to touch it to make it change (dial something, or just pick up and hangup) I couldnt find why this is happening, i searched, and found some with the same problem, but no solution, If you have any idea why this is happening, i will be glad to hear it. Thanks. Marco. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAXTel problems
Hi, I tried to add the IAXTel config to my asterisk, so i can dial free numbers inside the US from my SIP softphone (X-lite), everything seems to be working, but the sound quality is terrible, the other side sounds like a digitized voice, and the voice is cut, i cant hear a full word, I tried using FWD IAX interface, and no problem there, it works great. Now, although this is in a testing phase, i wanted to know if i am missing something, or IAXTel is just problematic . I am dialing from Israel, over a E1 line, dont know exactly how much of my E1 reaches the US, but should be sufficent for one session (for which FWD works fine with) Any help appriciated. Marco. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
