Re: [asterisk-users] sip realtime md5secret
OK. thanks for the replies. I missed the point about expiration, i will focus on this point. On Wed, Feb 03, 2010 at 08:58:00AM +0200, Mindaugas Kezys wrote: Just remember, that after reload you will lose all registrations. Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Chavez Sent: 2010 m. vasario 2 d. 22:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] sip realtime md5secret On Tue, 2010-02-02 at 21:20 +0200, Emre Kurnaz wrote: Hi all, Does asterisk cache realtime sip md5secret values? I create a user over a web site and set a password as asd and I can login with that password. After a while I change my password and set it as 123. Although the password is set as 123 in the mysql database (I double checked), i can not login using the password 123, but with asd. So, am i missing a point? or is this how asterisk works? and Should I reload asterisk after adding a peer in the database? Any help would be appreciated. If you have rtcachefriends=yes set in your sip.conf file then you either have to wait until the peer expires or you have to reload sip so the peer is re read from the database. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- 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 -- Emre Kurnaz ITU/BIDB Sistem Destek Grubu RHCE : 805009174841679 Yarı Zamanlı Öğrenci Koordinatörü kurn...@itu.edu.tr 0212 285 3930 ITU Linux Academy http://ila.itu.edu.tr -- _ -- 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] Intel Atom based Asterisk server?
On Wed, 3 Feb 2010, Remco Barendse wrote: I currently have some Asterisk home servers on general pc hardware as well as a mission critical server asterisk pbx running on a Dell 2850 To reduce noise and power consumption i would like to migrate them all to an Intel Atom based solution, showstoppers so far were single NIC and single PCI slot motherboards. I found that Supermicro makes a Dual NIC board with one PCI slot and 2 PCI-Express slots (X7SLA-L) Has anyone tried running Asterisk + CentOS 5 on this (or any other) Atom board? Is the Atom platform able to handle the load of all the interrupts a TE110P or TDM400P card will generate ? I am aware about other solutions but i do use the servers for some other tasks therefore don't want to move to a dedicated pbx box based on Soekris or the likes. I started on a 500Mhz VIA chip with TDM400 card and that coped (still does) very well with the call load of 12 people and 3 analogue lines... So anything bigger is not going to have any issues. I also have several other Atom based servers - Asterisk and otherwise. Beware the cheap (fast!) little fans on them though - every single one has failled on me so-far. (And this includes ones in clean air AC server room environments) Gordon -- _ -- 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] uri tel: instead of sip:accepted ?
3 feb 2010 kl. 08.11 skrev Alex Balashov: On 02/03/2010 02:03 AM, Olle E. Johansson wrote: 2 feb 2010 kl. 11.20 skrev BERGANZ Francois: Hello all, Does asterisk accept uri tel: instead of sip: ? No, but I think it would be a good addition. Why? Just curious. Well, adding a domain to a PSTN number doesn't make sense. In most cases, the PSTN number has nothing to do with the domain, the domain is just for routing which we do statically with outbound proxys and default routes anyway. To clearly separate PSTN phone numbers from SIP Uri's where the domain makes sense and the username part belongs to the domain makes more sense to me. I know that this is not how the current implementations out there are done, but I still feel that it would bring some order to the mess :-) /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] Asterisk 1.6.2 ?
On Wed, Feb 3, 2010 at 12:17 AM, Ben Dinnerville b...@voicelogic.com.auwrote: This is usually due to an error with the SIP stack not being loaded due to an error - make sure that full logging is on and check your log file and search for ERROR and see if there is any mention to SIP (chan_sip.o etc), alternatively, start asterisk from the command like with asterisk -vdc and watch the output to screen for any errors at startup. Fix the error and SIP will start up. -- _ -- 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 Thank you very much for your reply . I found my mistake . It was coming from my attempt to copy the old sip.conf extensions.conf onto the new build ones . It seems that it is not possible this way . -- _ -- 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] Pri HDLC aborts and choppy audio when dialling into pri, caused by BIOS option [CPU enhanced halt c1e]
Hardware: Digium TE110P REV.C and REV.D Gigabyte GA-965G-DS3 Bios F8b cat /proc/cpuinfo model name : Intel(R) Core(TM)2 CPU 6600 @ 2.40GHz stepping: 6 cpu MHz : 2400.080 cache size : 4096 KB ... latest libpri, dahdi, asterisk as of tonight. linux: debian lenny After moving hardware around all slots, disabling all unused hardware with no improvement finally disabled the Advanced Bios Option [CPU enhanced halt c1e] All choppy audio completely gone, no more HDLC aborts. Enable [CPU enhanced halt c1e] and it's all back. 100% repeatable. Could be related to more than just the above mentioned hardware. Alec Davis -- _ -- 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] CDR and Queue Reporting windows application looking for Beta testers!
Hi! I've been on this list for over 3 years and this is my first post. We have a reporting application for Asterisk that is soon to be in beta. It's a windows application that generates reports from log files (CDR and queue). It has a drag and drop approach to report creating. There is a pivot grid and you just put the data that you want in a report, where you want it. We're currently taking beta sign-ups. @ http://samreports.com Regards. Mihaela MJ. http://samreports.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
[asterisk-users] Asterisk core sounds in English by June Wallack
Is there a version of the Asterisk core sounds in English done by June Wallack? Some folks here prefer her voice to Allison's, but we'd like consistency. And is there a version of the Cepstral software with her voice? -- _ -- 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] Semi-Transfer
I've tried that as well prior to sending the initial email with no
results.
I'll play some with DISA today.
James Shigley
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
Nicholas
Sent: Tuesday, February 02, 2010 2:50 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Semi-Transfer
This wiki is outdated but the group stuff still applies to DAHDI
http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels
Assuming that you have many available lines in group 3, changing the
option to g3 from G3 might help.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James A.
Shigley
Sent: Tuesday, February 02, 2010 2:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Semi-Transfer
That is the PRI span there are many available lines.
James Shigley
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
Nicholas
Sent: Tuesday, February 02, 2010 2:12 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Semi-Transfer
What lines are in your group 3? It is possible that DAHDI/52 is the
only line in that group and that's why you're getting the all
congested.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James A.
Shigley
Sent: Tuesday, February 02, 2010 2:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Semi-Transfer
There are times when I need to call a client from my cell and I want a
recording of the call. I was trying to put into an * a way of doing
that. Below is what I'm using in my extensions.conf
exten= X,1,Read(num,/var/lib/asterisk/sounds/mtas/10digit,10,,,5)
exten= X,2,SayDigits(${num})
exten= X,3,Background(/var/lib/asterisk/sounds/mtas/verify)
exten= X,4,WaitExten(3)
exten=
X,5,Monitor(wav,/var/store/calls/InOutRec-${STRFTIME(${EPOCH},,%Y%m%d-%H
%M%S)}-${CALLERID(num)}-${EXTEN},mb)
exten= X,6,dial(${belltd}/${num})
Here is what I see in the CMD when the dial fails
-- Timeout on DAHDI/52-1, continuing...
-- Executing [xxx...@recout:5] Monitor(DAHDI/52-1,
wav,/var/store/calls/InOutRec-20100202-133012-XX-XX,mb
) in new stack
-- Executing [XX @RecOut:6] Dial(DAHDI/52-1,
DAHDI/G3/4099819750) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called G3/4099819750
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'DAHDI/52-1' status is 'CHANUNAVAIL'
-- Hungup 'DAHDI/52-1'
Now all of my lines are NOT ties up so there is available paths for the
call to go out.
Anyway so how would I accomplish this transfer of sorts?
James Shigley
Monroe Telephone Answering Service
409-981-9750
Infinity 5.51,UC 4.02.3803, Blink 3.0.104
Ecreator:2.21, eResponse 1.1.7
Webportal,WebApps,
CONFIDENTIALITY NOTICE: This email, including any attachments, contains
information which may be confidential or privileged. The information =is
intended to be for the use of the individual or entity named above. If
you are not the intended recipient, be aware that any disclosure,
copying, distribution or use of the contents of this information is
prohibited. If you have received this email in error, please notify the
sender immediately by reply to sender only message and destroy all
electronic and hard copies of the communication, including attachments.
Side Note: I am James, Jim is my future father in law!
image001.jpg--
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[asterisk-users] ast_cdr_setvar: Attempt to set the 'src' read-only variable!
Hello list. I would like to set the CDR(src)-variable to the SIPphone that is initiating the call. When calling out, the src-variable is always the public telephone number. I get the ERROR : ast_cdr_setvar: Attempt to set the 'src' read-only variable! Is there some way to implement this ?? Kind regards, Jonas. -- _ -- 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] Asterisk core sounds in English by June Wallack
Richard Kenner wrote: Is there a version of the Asterisk core sounds in English done by June Wallack? Some folks here prefer her voice to Allison's, but we'd like consistency. And is there a version of the Cepstral software with her voice? No, neither of those exist, and I'm not aware of any plans to produce them. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com 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] ast_cdr_setvar: Attempt to set the 'src' read-only variable!
On Wednesday 03 February 2010 09:24:34 jonas kellens wrote: I would like to set the CDR(src)-variable to the SIPphone that is initiating the call. When calling out, the src-variable is always the public telephone number. I get the ERROR : ast_cdr_setvar: Attempt to set the 'src' read-only variable! Is there some way to implement this ?? Set(CALLERID(ani)=whatyouwantinsrc) -- Tilghman -- _ -- 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] Intel Atom based Asterisk server?
At 11:19 PM 2/2/2010, you wrote: Has anyone tried running Asterisk + CentOS 5 on this (or any other) Atom board? Is the Atom platform able to handle the load of all the interrupts a TE110P or TDM400P card will generate ? I run 3 pots lines into a TDM04 on an Atom 330 board with no problems. 3 POTS and 4 SIP lines coming in, 3 Aastra SIP phones and 2 people. Been running the Atom for a year or so. Building takes a bit longer than the old machine, but not enough to be annoying. Ira -- _ -- 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] Intel Atom based Asterisk server?
Ira, Are you running anything clunky on there like FreePBX or mysql? And based on your experience with the Atom board, do you think you would have performance problems with said clunkiness? Thanks, Lyle On Wed, 2010-02-03 at 09:50 -0800, Ira wrote: At 11:19 PM 2/2/2010, you wrote: Has anyone tried running Asterisk + CentOS 5 on this (or any other) Atom board? Is the Atom platform able to handle the load of all the interrupts a TE110P or TDM400P card will generate ? I run 3 pots lines into a TDM04 on an Atom 330 board with no problems. 3 POTS and 4 SIP lines coming in, 3 Aastra SIP phones and 2 people. Been running the Atom for a year or so. Building takes a bit longer than the old machine, but not enough to be annoying. Ira -- _ -- 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] Intel Atom based Asterisk server?
The Atom boards are fine for Asterisk and even some clunkyness... Transcoding is where the Atom boards could cause issues but this is in the realm of 50 channels or so... ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Wed, Feb 3, 2010 at 3:28 PM, Lyle Underwood lyleunderw...@gmail.com wrote: Ira, Are you running anything clunky on there like FreePBX or mysql? And based on your experience with the Atom board, do you think you would have performance problems with said clunkiness? Thanks, Lyle On Wed, 2010-02-03 at 09:50 -0800, Ira wrote: At 11:19 PM 2/2/2010, you wrote: Has anyone tried running Asterisk + CentOS 5 on this (or any other) Atom board? Is the Atom platform able to handle the load of all the interrupts a TE110P or TDM400P card will generate ? I run 3 pots lines into a TDM04 on an Atom 330 board with no problems. 3 POTS and 4 SIP lines coming in, 3 Aastra SIP phones and 2 people. Been running the Atom for a year or so. Building takes a bit longer than the old machine, but not enough to be annoying. Ira -- _ -- 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] Intel Atom based Asterisk server?
Just Asterisk 1.6.2 on CENTOS 5. I can't imagine any problems at all, I don't think top has ever been over a couple percent except when I'm re-building asterisk. It's got 2Gb of ram so everything is in ram, I've never seen the swap file usage get past zero. Right now there's 512K free so it's a long way to go before hitting the disk. Ira At 10:28 AM 2/3/2010, you wrote: Ira, Are you running anything clunky on there like FreePBX or mysql? And based on your experience with the Atom board, do you think you would have performance problems with said clunkiness? I run 3 pots lines into a TDM04 on an Atom 330 board with no problems. 3 POTS and 4 SIP lines coming in, 3 Aastra SIP phones and 2 people. Been running the Atom for a year or so. Building takes a bit longer than the old machine, but not enough to be annoying. Ira -- _ -- 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] calling into server with cell and originate a call
Hi All, Can anyone tell me how I could originate a call from my server? My use case is I am on the road and I want to dial into the server using my cell phone, log into my user account and then enter a number to dial so that the call comes from the server. Thanks, John -- _ -- 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] Intel Atom based Asterisk server?
- Ira i...@extrasensory.com wrote: Just Asterisk 1.6.2 on CENTOS 5. I can't imagine any problems at all, I don't think top has ever been over a couple percent except when I'm re-building asterisk. It's got 2Gb of ram so everything is in ram, I've never seen the swap file usage get past zero. Right now there's 512K free so it's a long way to go before hitting the disk. Ira At 10:28 AM 2/3/2010, you wrote: Ira, Are you running anything clunky on there like FreePBX or mysql? And based on your experience with the Atom board, do you think you would have performance problems with said clunkiness? I run 3 pots lines into a TDM04 on an Atom 330 board with no problems. 3 POTS and 4 SIP lines coming in, 3 Aastra SIP phones and 2 people. Been running the Atom for a year or so. Building takes a bit longer than the old machine, but not enough to be annoying. Ira top - 19:49:55 up 1:27, 1 user, load average: 0.00, 0.00, 0.00 Tasks: 122 total, 1 running, 121 sleeping, 0 stopped, 0 zombie Cpu(s): 0.3%us, 0.3%sy, 0.0%ni, 98.8%id, 0.2%wa, 0.4%hi, 0.0%si, 0.0%st Mem: 2065860k total, 591552k used, 1474308k free,44412k buffers Swap: 4128760k total,0k used, 4128760k free, 443504k cached PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 2937 root -11 0 2 22m 11m S 1.9 1.1 0:17.02 asterisk 11811 root 15 0 2360 920 696 R 1.9 0.0 0:00.01 top Uptime low as just rebooted for AST-2010-001. CDR is on a MySQL instance on the same server. All SIP channels apart from DAHDI on TDM400p for backup purposes. -- Thanks, Phil -- _ -- 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] calling into server with cell and originate a call
On 3 Feb 2010, at 19:17, John Regal wrote: Can anyone tell me how I could originate a call from my server? My use case is I am on the road and I want to dial into the server using my cell phone, log into my user account and then enter a number to dial so that the call comes from the server. DISA -- _ -- 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] calling into server with cell and originate a call
Thanks! That was easy. :) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: Wednesday, February 03, 2010 3:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] calling into server with cell and originate a call On 3 Feb 2010, at 19:17, John Regal wrote: Can anyone tell me how I could originate a call from my server? My use case is I am on the road and I want to dial into the server using my cell phone, log into my user account and then enter a number to dial so that the call comes from the server. DISA -- _ -- 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] 500 Internal Server Error on Cisco 7940 after INVITE
Hi, Our Cisco 7940 phones on a single network sometimes seem to drop calls as soon as they are picked up. After a second INVITE the phone sends '500 Internal Error'. One phone thinks its still in a call (though there is no audio) while the other phone is not in a call. The drop happens immediately after connecting and I think it is due to a second INVITE. Turning reinvite off seems like it fixes the problem but I haven't tested enough to be sure yet. The bug only seems to happen on the first time a call is made after a period of time (and even then only occasionally). Successive calls between the same phones are always fine. That makes debugging it a bit of a pain. I have managed to capture a packet trace between * (10.200.4.100) and the 7940 (10.200.4.66) showing the error. 36.437814 10.200.4.100 - 10.200.4.66 SIP/SDP Request: INVITE sip:3...@10.200.4.66:5060;transport=udp, with session description 36.584594 10.200.4.66 - 10.200.4.100 SIP Status: 100 Trying 36.720893 10.200.4.66 - 10.200.4.100 SIP Status: 180 Ringing 43.211744 10.200.4.66 - 10.200.4.100 SIP/SDP Status: 200 OK, with session description 43.212001 10.200.4.100 - 10.200.4.66 SIP Request: ACK sip:3...@10.200.4.66:5060;transport=udp 43.212536 10.200.4.100 - 10.200.4.66 SIP/SDP Request: INVITE sip:3...@10.200.4.66:5060;transport=udp, with session description 43.304295 10.200.4.66 - 10.200.4.100 SIP Status: 500 Internal Server Error 43.304489 10.200.4.100 - 10.200.4.66 SIP Request: ACK sip:3...@10.200.4.66:5060;transport=udp 43.402253 Cisco_76:a5:0f - BroadcastARP Who has 10.200.4.100? Tell 10.200.4.66 43.470520 10.200.4.66 - 10.200.4.100 RTP PT=ITU-T G.711 PCMA, SSRC=0xFA576B7, Seq=2616, Time=414864, Mark There is a description of an error http://lists.iptel.org/pipermail/serdev/2005-November/006344.html which seems similar but since its in 2005 I assume the problem would be fixed by now. If that is the problem, can anybody suggest a workaround? I have upgraded the phones to the most recent firmware (POS3-08-11-00) and * is Version: 1:1.4.21.2~dfsg-3+lenny1 (debian). Can anybody offer any advice or suggestions on fixing or debugging this further? If anybody else has encountered the problem and knows of a fix that would be great. Thanks, Ian Crowther -- === Ian Crowther Tel: +44 845 4501626 Unit 108, 10th Avenue, IT Dept, ComtekFax: +44 845 4501627 Zone 3, Deeside Industrial Network Systems UK Ltd Park, CH5 2UA, Flintshire === -- _ -- 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] Dial cellphone from one PBX1 to PBX2? is it possible?
OK! you're right Mr William, that worked, unfortunately i don't know if
there is any points to affect to somebody who helped you (like rating in
other forums) :(, but one more request can you recommand me a book to master
and tweak dialplans?
2010/1/29 William Stillwell (Lists) william.stillwell-li...@ablebody.net
You are using contexts..
Look @ destination pbx, you should see something like this:
Rejected connect attempt from ip of source pbx, request ‘ext@incoming
conext’ does not exist
If you didn’t put a context under the peer, it uses the default one in the
iax.conf file which is normally [default]
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *khalid touati
*Sent:* Friday, January 29, 2010 11:54 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Dial cellphone from one PBX1 to PBX2? is
it possible?
Hi William,
I appreciate your answer, though can you make things more clear for me:
1- i am not using extensions when registering PBX boxes in IAX files.
2- is inbounx context in the call sender PBX (pbx1) and outbound context is
in the call receiver (or dialer) PBX (pbx2)?
3- i am using two identical dialplan's is this gonna confuse
the communication process (contextes's name are duplicated over the two
servers)
thank you very much for making it clear for me!
2010/1/28 William Stillwell (Lists) william.stillwell-li...@ablebody.net
Your inbound context needs to have access to your outbound context.
[iax-inbound]
Include = outbound-conext
[outbound-context]
Exten = _1NXXNXX,1,Dial(DAHDI\g1\${EXTEN})
Something like that.
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *khalid touati
*Sent:* Thursday, January 28, 2010 3:29 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Dial cellphone from one PBX1 to PBX2? is it
possible?
Hi Guys,
i am using two PBX's i can call from pbx1 a cellphone tied to pbx2 that
way:
1) use a phone in PBX1
2) call extension in PBX2
3) extensions in PBX2 ring to a cellphone (as this specific ext is tied to
a cellphone)
my questions now is : am i gonna be able to dial from an IPphone registered
within PBX1 to a cellphone by using the Trunk (Zap or dahdi) of PBX2?
anybody know
IPphone-PBX1-IAXPBX2PRI
line---cellphone???
thank you for you help guys!!
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Re: [asterisk-users] 500 Internal Server Error on Cisco 7940 after INVITE
snip I have upgraded the phones to the most recent firmware (POS3-08-11-00) and * is Version: 1:1.4.21.2~dfsg-3+lenny1 (debian). snip That doesn't look like cisco firmware to me... Unless I'm mistaken. What version are the phones on? (Settings = Status = Firmware Versions) -Dave -- _ -- 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] Routing inbound call to correct sip trunk
There's some one who can help me? I'm using Asterisknow with FreePBX and a Patton 4554 with 2 BRI ports on 2 ISDN lines. I would like routing the call entering by first BRI to one trunk and call from second BRI to another trunk. I have created 2 trunks both registering to Patton with different identities, actually all calls from both BRI are routed to one trunk. Thank in advance -- _ -- 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] aastra 9480i dtmf ?
Hi, I just deployed new Aastra 9480i phones and when I attempt enter digits on other systems, like host pin in a GoToMeeting, the servers on the other end do not get my entries. I am assuming this is a DTMF issue but do not see anything in this phones config other than turning on the display of the digits. I have the DTMF method set to SIP INFO. I am using AsteriskNow w/FreePBX. Thanks in advance. -- _ -- 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] aastra 9480i dtmf ?
It appears I am having this same problem with softphones, too. It seems that if I end my request by pressing # twice(slowly) as opposed to once as requested by the system, it works. From: John Regal [mailto:jre...@gmail.com] Sent: Wednesday, February 03, 2010 5:25 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: aastra 9480i dtmf ? Hi, I just deployed new Aastra 9480i phones and when I attempt enter digits on other systems, like host pin in a GoToMeeting, the servers on the other end do not get my entries. I am assuming this is a DTMF issue but do not see anything in this phones config other than turning on the display of the digits. I have the DTMF method set to SIP INFO. I am using AsteriskNow w/FreePBX. Thanks in advance. -- _ -- 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] 500 Internal Server Error on Cisco 7940 after INVITE
David Gibbons wrote: snip I have upgraded the phones to the most recent firmware (POS3-08-11-00) and * is Version: 1:1.4.21.2~dfsg-3+lenny1 (debian). snip That doesn't look like cisco firmware to me... Unless I'm mistaken. What version are the phones on? (Settings = Status = Firmware Versions) -Dave Thats straight out of that section. Its the most recent SIP firmware I could find. Application Load ID: 'POS3-08-11-00'. Boot Load ID: PC03A300 DSP Load ID 4.0(5.0)[A0] It seems to mean 8.11. http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/7960g_7940g/firmware/sip/8_11/english/release/notes/796040sip_811.html#wp1099767 Thanks, Ian -- === Ian Crowther Tel: +44 845 4501626 Unit 108, 10th Avenue, IT Dept, ComtekFax: +44 845 4501627 Zone 3, Deeside Industrial Network Systems UK Ltd Park, CH5 2UA, Flintshire === -- _ -- 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] 1.6.2.1: DTMF trouble with PSTN
Using 1.6.2.1 with a TDM400, attached to internal analog phones and PSTN. When I dial out to PSTN, I cannot send tones, like push 1 for something stupid. The call itself works, but the DTMF tones fail. -- Starting simple switch on 'DAHDI/1-1' -- Executing [6258...@internal:1] Answer(DAHDI/1-1, ) in new stack -- Executing [6258...@internal:2] Dial(DAHDI/1-1, DAHDI/4/ww2156258013) in new stack -- Called 4/ww2156258013 -- DAHDI/4-1 answered DAHDI/1-1 -- Native bridging DAHDI/1-1 and DAHDI/4-1 -- Hungup 'DAHDI/4-1' Any suggestions? sean -- _ -- 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] Astribank problem
Ok, the problem solved... thanks for your advice. after rebooting i run /usr/share/dahdi/xpp_fxloader load and everything run normally. thanks... Message: 24 Date: Mon, 1 Feb 2010 11:03:44 +0200 From: Tzafrir Cohen tzafrir.co...@xorcom.com Subject: Re: [asterisk-users] Astribank problem To: asterisk-users@lists.digium.com Message-ID: 20100201090344.gr3...@xorcom.com Content-Type: text/plain; charset=us-ascii On Mon, Feb 01, 2010 at 07:42:51AM +, frangky robert wrote: I do some test: 1.unplug usb connector from server to astricon 2.unplug power to astricon 3.plug-in the power to astricon 4.plug-in the usb connector Here is the log from /var/log/messages after doing the 1st step. Feb 1 19:38:24 localhost last message repeated 2 times Feb 1 19:43:39 localhost kernel: ERR-xpp_usb: xusb-0 (usb-:00:1d.7-3) [X1038295]: nonzero write bulk status received: -71 (pending_writes=1) Feb 1 19:43:39 localhost kernel: usb 2-3: USB disconnect, address 3 Feb 1 19:43:39 localhost kernel: ERR-xpp_usb: XBUS-00: xusb_listen: usb_submit_urb failed: -19 Feb 1 19:43:39 localhost kernel: INFO-xpp: XBUS-00: [usb:X1038295] Disconnecting Feb 1 19:43:39 localhost kernel: INFO-xpp: XBUS-00: [usb:X1038295] Deactivating Feb 1 19:43:39 localhost kernel: INFO-xpp: XBUS-00: [usb:X1038295] Release XPDS Feb 1 19:43:39 localhost kernel: INFO-xpp: XBUS-00/XPD-00: Remove Feb 1 19:43:39 localhost kernel: INFO-xpp: XBUS-00/XPD-10: Remove Feb 1 19:43:39 localhost kernel: INFO-xpp: XBUS-00/XPD-20: Remove Feb 1 19:43:39 localhost kernel: NOTICE-xpp: worker_reset: worker(XBUS-00)-xpds_init_done=0 Feb 1 19:43:39 localhost kernel: INFO-xpp: XBUS-00: [usb:X1038295] Atribank Remove Feb 1 19:43:39 localhost kernel: INFO-xpp: XBUS-00: [usb:X1038295] Astribank Release Feb 1 19:43:39 localhost kernel: INFO-xpp_usb: xusb-0 (usb-:00:1d.7-3) [X1038295]: now disconnected Feb 1 19:43:39 localhost 'astribank_hook'[3728]: offline(XBUS-00): 0/1 from /etc/dahdi/xpp_order Feb 1 19:43:39 localhost 'astribank_hook'[3735]: All Astribanks offline And, this is the log after doing 4th step. Feb 1 19:44:20 localhost kernel: usb 2-3: new high speed USB device using ehci_hcd and address 4 Feb 1 19:44:20 localhost kernel: usb 2-3: configuration #1 chosen from 1 choice Feb 1 19:44:21 localhost 'xpp_fxloader'[3847]: Exiting... XPP_HOTPLUG_DISABLED Seems like you explicitly disabled firmware loading by setting XPP_HOTPLUG_DISABLED in /etc/dahdi/init.conf . Just rem-out that line. lsusb result is: [r...@localhost ~]# lsusb Bus 002 Device 004: ID e4e4:1160 Bus 002 Device 001: ID : Bus 006 Device 001: ID : Bus 006 Device 002: ID 04b3:3025 IBM Corp. Bus 004 Device 001: ID : Bus 008 Device 001: ID : Bus 007 Device 001: ID : Bus 001 Device 001: ID : Bus 005 Device 001: ID : Bus 003 Device 001: ID : here is the msg when i do /usr/share/dahdi/xpp_fxloader [r...@localhost ~]# /usr/share/dahdi/xpp_fxloader usb This only runs USB firmware loading. And as the firmware loading is explicitly disabled on your system, the FPGA firmware will still not get loaded. This is also something that you would have seen if you would run 'dahdi_hardware -v' So basically just remove that line from init.conf and replug the Astribank. -- 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 * _ New Windows 7: Find the right PC for you. Learn more. http://windows.microsoft.com/shop-- _ -- 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
