Re: [asterisk-users] New tutorial: storing audio recordings per day
Thank you! I updated the tutorial as well.
l.
2009/5/25 Atis Lezdins a...@iq-labs.net
On Mon, May 25, 2009 at 7:42 PM, Lenz Emilitri lenz.lo...@gmail.com
wrote:
Hi everyone,
after doing the same thing multiple times and struggling to remember how
it
was done, I have prepared a small tutorial that explains how to save
monitored files in different folders per day. This is quite useful
becausethe resultingfile system is way more manageable than having maybe
100,000 files all saved in the same folder.
You can find the tutorial here:
http://astrecipes.net/index.php?n=387
As always, comments and suggestions are welcome.
l.
PS. I am also working on some scripts to normalize existing recordings
all-in-one-directory... if anybody is interested, please contact me.
Actually You don't have to create folders in advance, as Asterisk will
automatically create them when needed. Just make sure that Asterisk
process is owner of parent directory.
Set(__call_day=${STRFTIME(|${TIMEZONE}|%Y/%m/%d)});
Set(MONITOR_FILENAME=${MONITOR_DIR}/${call_day}/call-${UNIQUEID});
Monitor(ulaw,${MONITOR_FILENAME},b);
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
a...@iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
___
-- 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
--
Loway - home of QueueMetrics - http://queuemetrics.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-addon 1.6.1 problem
Hi all, I download asterisk-addon 1.6.1 but the VoIP phone failed to register to the system with the message below. [May 26 15:45:11] WARNING[29665]: res_config_mysql.c:317 realtime_mysql: MySQL RealTime: Invalid database specified: asterisk [May 26 15:45:11] WARNING[29665]: res_config_mysql.c:317 realtime_mysql: MySQL RealTime: Invalid database specified: asterisk sip I use the same configuration file (res_mysql.conf extconfig.conf) in 1.6.0 but failed. Any big change in 1.6.1? ___ -- 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] h extension and channel variables
Is there a method to fetch the ${EXTEN} of the channel that has been
hung up when exten h is started?
The nearest thing I can think of is to set another variable to the
extension and pick that up. Would that be a reliable method though?
___
-- 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 and Data Modem
Hi All, We have an Asterisk 1.4.21.2 box which uses a 2 port Digium card to bridge calls, as follows: ISDN Provider --- Span 1(pri_cpe) --- Span 2(pri_net) Phone System The company that looks after our internal phone system can no longer dial in using their data modem in order to maintain the internal phone system. Is there any way we can configure our asterisk to allow them to dial in using their modem? Regards, Jon. ___ -- 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] RDNIS question
Hi I am a premium voice service provider giving some services on IVR to a Telco X . As my premises is some 10 kms away from that telco , i have taken a PRI connection (30 DID with 1 hunting/pilot number) from telco Y When a customer of Telco X dials my short code @Rs.6/- per minute his call is forwarded on the PRI connection of telco Y . All this works fine.. Now the problem arises during billing , many customers of Telco X / Telco Z / Telco Y somehow get to know the pilot number of telco Y and they directly dial in (it becomes a local call and not a premium rate) the rsult being i dont get paid for those minutes and am giving the service free virtually ...I tried to solve the problem as follows : 1. If i filter the calls using DNIS - no matter people call short code or my pilot number - the DNIS would always be returned as the pilot number 2. If i filter calls using ANI so that i allow only customer of Telco X , then eventhough i minimise the damage - but still am not sure if that customer X has dialled short code or long code ? 3. Can RDNIS function help me in anyway ? this question may sound off-topic but in asterisk is there a way out ? Rgds sriram___ -- 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 and Data Modem
Sure - you just need to figure out what number is being dialed, make sure that number rings on the incoming PRI, and make sure the phone system expects that call to come in the standard PRI trunk group and not some dedicated analog craft port. -- Sent from mobile device On May 26, 2009, at 6:05 AM, Jon Morgan jmor...@c-a- solutions.co.uk wrote: Hi All, We have an Asterisk 1.4.21.2 box which uses a 2 port Digium card to bridge calls, as follows: ISDN Provider --- Span 1(pri_cpe) --- Span 2(pri_net) Phone System The company that looks after our internal phone system can no longer dial in using their data modem in order to maintain the internal phone system. Is there any way we can configure our asterisk to allow them to dial in using their modem? Regards, Jon. ___ -- 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] RDNIS question
On 26 May 2009, at 11:48, Sriram wrote: Now the problem arises during billing , many customers of Telco X / Telco Z / Telco Y somehow get to know the pilot number of telco Y and they directly dial in. How exactly? You might have it accidently listed somewhere. Worth just looking on online phone directories etc. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] h extension and channel variables
On 5/26/2009 10:57, Thomas Kenyon wrote:
Is there a method to fetch the ${EXTEN} of the channel that has been
hung up when exten h is started?
The nearest thing I can think of is to set another variable to the
extension and pick that up. Would that be a reliable method though?
Which is clearly a bad idea, since an intervening call would change this.
My Best idea so far is to change the CallerID to the exten (although it
may be desirable to keep it in tact, it's not as important in this case).
Does anybody have any suggestions?
___
-- 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] Logging calls made/lost
Hi, I'm in the process for setting up an asterisk server for four organisations sharing a SIP trunk. In order to split the costs according to usage, it would be nice to log all incoming, outgoing and missed calls. Is there a simple way of doing this, preferrably in a database? Perhaps someone has made a solution with a simple web interface already? Any suggestions would be welcome :-) -- Andreas-Johann Ulvestad Dagleg leiar, Unicornis ___ -- 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] Domains
Hi, I'm trying to understand an issue I'm seeing between two Asterisk servers. I think it has to do with Domain definitions. Server A), has extension 5550 defined. Has a sip client 2000 defined, and has guest-invites enabled. Server B), Dials to server A for any 5550 dialled. Has sip client 2000 and 2001 defined. If I register at server B as client 2001, and dial 5550 then the call works, and is placed through to server As logic successfully. But if I call in as client 2000, then the call fails, server A shows no log at all of the call (even a sip set debug ip ip showed nothing - though tcpdump did show the inbound invite). However if I remove the definition of client 2000 from server A, then the call succeeds. So I think that for a defined account server A is wanting to challenge for a password, even though the inbound call is not a local account - hence my trying now to understand if and how Asterisk uses Domains. If I define a serverA.company.com domain on server A, will it ignore the challenge for an INVITE coming from server B ?? Thanks Adrian ___ -- 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] Bandwidth management and ADSL router
Hi All; I discover that most of the voice cutting complain are coming from the Internet bandwidth when we are connecting two remote offices togethor via Asterisk or any other IP PBX. Anyone has an idea on a ADSL router that work as ADSL + Bandwidth division? So we can resolve the problem of providing a guaranteed bandwidth for the voice packets instead of suffering the voice cutting? Regards Bilal ___ -- 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] Converting Cisco 7961 to SIP
As part of a project to move a clients Cisco phones to SIP to work with an Avaya system, I'm taking a Cisco 7961 we bought and adding it to our Asterisk setup. Now, I've gotten the firmware files from the site, the latest version is 8.4 which contains the following files: apps41.8-4-3-16.sbn cnu41.8-4-3-16.sbn cvm41sip.8-4-3-16.sbn dsp41.8-4-3-16.sbn jar41sip.8-4-3-16.sbn SIP41.8-4-4S.loads term41.default.loads term61.default.loads Now I've read over loads of documentation on it, but am getting tripped up. Most of what I've seen talks about the older firmware versions usually 7.4. I have a feeling I'm still missing a lot of stuff. Anyone have any recent links or information? Also, anyone know of a decent way to generate the config files? I'd hate to have to go through all of it manually? Thanks. Cheers, Darrin Henshaw This email and its attachments may be confidential and are intended solely for the use of the individual or parties' to whom it is addressed. All comments are solely those of the author and do not necessarily represent those of Ignition. If you are not the intended recipient of this email and its attachments, you must take no action based upon them, nor must you copy or show them to anyone. Please contact the sender if you believe you have received this email in error. Thanks for considering the environmental impact before printing this email. ___ -- 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] Bandwidth management and ADSL router
A lot of the ADSL CPE (customer premise equipment) deployed has basic QoS capabilities in a pre-set kind of way, but if you want to do your own DiffServ tagging the standard practice is to do Layer 2 Ethernet bridging to a more intelligent box behind the ADSL CPE. bilal ghayyad wrote: Hi All; I discover that most of the voice cutting complain are coming from the Internet bandwidth when we are connecting two remote offices togethor via Asterisk or any other IP PBX. Anyone has an idea on a ADSL router that work as ADSL + Bandwidth division? So we can resolve the problem of providing a guaranteed bandwidth for the voice packets instead of suffering the voice cutting? Regards Bilal ___ -- 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- 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] Bandwidth management and ADSL router
m0n0wall and pfsense both do traffic shaping, which forcibly allocates bandwidth for your VoIP traffic. Michael On Tue, 26 May 2009 04:32:59 -0700 (PDT), bilal ghayyad wrote: Hi All; I discover that most of the voice cutting complain are coming from the Internet bandwidth when we are connecting two remote offices togethor via Asterisk or any other IP PBX. Anyone has an idea on a ADSL router that work as ADSL + Bandwidth division? So we can resolve the problem of providing a guaranteed bandwidth for the voice packets instead of suffering the voice cutting? Regards Bilal ___ -- 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 -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves fwd 54245 ___ -- 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] Bandwidth management and ADSL router
On Tue, 26 May 2009, bilal ghayyad wrote: Hi All; I discover that most of the voice cutting complain are coming from the Internet bandwidth when we are connecting two remote offices togethor via Asterisk or any other IP PBX. Anyone has an idea on a ADSL router that work as ADSL + Bandwidth division? So we can resolve the problem of providing a guaranteed bandwidth for the voice packets instead of suffering the voice cutting? Draytek 2800 series routers have adequate traffic management on outgoing traffic to do a reasonable job. (There is a very little you can do to shape incoming traffic) However you need to make sure that the actual Internet connection isn't where the bottleneck is. Try making calls when you can guarantee that no other traffic is flowing into/out of each end. 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] Bandwidth management and ADSL router
As does ZeroShell (www.zeroshell.net/eng). Bruce Komito WPTI Telecom (775) 236-5815 On Tue, 26 May 2009, Michael Graves wrote: m0n0wall and pfsense both do traffic shaping, which forcibly allocates bandwidth for your VoIP traffic. Michael On Tue, 26 May 2009 04:32:59 -0700 (PDT), bilal ghayyad wrote: Hi All; I discover that most of the voice cutting complain are coming from the Internet bandwidth when we are connecting two remote offices togethor via Asterisk or any other IP PBX. Anyone has an idea on a ADSL router that work as ADSL + Bandwidth division? So we can resolve the problem of providing a guaranteed bandwidth for the voice packets instead of suffering the voice cutting? Regards Bilal ___ -- 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 -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves fwd 54245 ___ -- 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] Hanging up a call by DTMF
Hello, Is it possible to hangup an active call by simply sending a DTMF code to Asterisk for example # code. If yes, What function to use in the dialplan. Thanks. ___ -- 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] h extension and channel variables
I set a variable CalledID to ${EXTEN} before dial it. So in h extension I
can use ${CalledID}.
2009/5/26 Thomas Kenyon dig...@sanguinarius.co.uk
On 5/26/2009 10:57, Thomas Kenyon wrote:
Is there a method to fetch the ${EXTEN} of the channel that has been
hung up when exten h is started?
The nearest thing I can think of is to set another variable to the
extension and pick that up. Would that be a reliable method though?
Which is clearly a bad idea, since an intervening call would change this.
My Best idea so far is to change the CallerID to the exten (although it
may be desirable to keep it in tact, it's not as important in this case).
Does anybody have any suggestions?
___
-- 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] Unable to make outbound calls
Sorry. I don't get many opportunities to test this system as its live. Here are the results: -- Executing [...@dlpn_dialplan1:1] Dial(SIP/19722-b650fb80, DAHDI/1) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called 1 -- Channel 0/1, span 1 got hangup, cause 90 WARNING[6454]: app_dial.c:765 wait_for_answer: Unable to forward voice or dtmf -- Hungup 'DAHDI/1-1' == Everyone is busy/congested at this time (1:0/0/1) On 20/5/09 3:52 PM, Danny Nicholas da...@debsinc.com wrote: This all looks ok. What happens if you try to access the DAHDI channel outside of Asterisk control: In dialplan Exten = 9,1,Dial(DAHDI/1) Dial 9 Get dialtone Dial number -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kal Feher Sent: Wednesday, May 20, 2009 2:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Unable to make outbound calls I attached the show channels in my first post, but removed it to reduce the length of replies. Here it is again along with show status. Note that there is only 1 PRI currently attached. geriatrix*CLI dahdi show status Description Alarms IRQbpviol CRC4 T2XXP (PCI) Card 0 Span 1OK 0 0 0 T2XXP (PCI) Card 0 Span 2RED0 0 0 geriatrix*CLI dahdi show channels Chan Extension Context Language MOH Interpret pseudodefaultdefault 1DID_span_1 default 2DID_span_1 default 3DID_span_1 default 4DID_span_1 default 5DID_span_1 default 6DID_span_1 default 7DID_span_1 default 8DID_span_1 default 9DID_span_1 default 10DID_span_1 default 11DID_span_1 default 12DID_span_1 default 13DID_span_1 default 14DID_span_1 default 15DID_span_1 default 16DID_span_1 default 17DID_span_1 default 18DID_span_1 default 19DID_span_1 default 20DID_span_1 default 21DID_span_1 default 22DID_span_1 default 23DID_span_1 default 25DID_span_2 default 26DID_span_2 default 27DID_span_2 default 28DID_span_2 default 29DID_span_2 default 30DID_span_2 default 31DID_span_2 default 32DID_span_2 default 33DID_span_2 default 34DID_span_2 default 35DID_span_2 default 36DID_span_2 default 37DID_span_2 default 38DID_span_2 default 39DID_span_2 default 40DID_span_2 default 41DID_span_2 default 42DID_span_2 default 43DID_span_2 default 44DID_span_2 default 45DID_span_2 default 46DID_span_2 default 47DID_span_2 default On 19/5/09 6:31 PM, Danny Nicholas da...@debsinc.com wrote: Please post your CLI output from dahdi show status and dahdi show channels. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kal Feher Sent: Tuesday, May 19, 2009 11:22 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Unable to make outbound calls Can you make an outbound call after your first incoming call? That's what I experienced with 1.4.24. No I can't. I saw similar threads while searching for a solution, but in my case outbound calls are never possible. Regardless, thank you for your suggestion. On 19/5/09 4:47 PM, Kal Feher kalman.fe...@melbourneit.com.au wrote: I've got an asterisk 1.4.24 box with dahdi
Re: [asterisk-users] Hanging up a call by DTMF
If you do Dial(tech/line,,Hh), either side can hang up the call with *. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of abdelkader Sent: Tuesday, May 26, 2009 7:46 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Hanging up a call by DTMF Hello, Is it possible to hangup an active call by simply sending a DTMF code to Asterisk for example # code. If yes, What function to use in the dialplan. Thanks. ___ -- 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] Converting Cisco 7961 to SIP
Darrin, The files you are using are consistent with SIP for Cisco Call Manager. Anything other than Callmanager will essentially be a hack. I am not sure how proprietary the Avaya system is in regards to registration and open-SIP support. Asterisk and any iteration of it will support it, but Cisco hasn't really designed a load compatible with it yet. I can tell you that I haven't really found any configuration file generation tools for these files. The reason being is that these loads are mainly used for SCCP and SIP Cisco systems. There is a well known tutorial on how to Hack to the CP-7970 to trixbox CE located here: http://www.asterisktutorials.com/cisco-7970-ip-phone/ This may help get you jump started and pointed in the right direction. The only problem that may arise is that in the tutorial, the use a specific SIP load (8.0.3) which may not be available for the 7961G. Cory J. Andrews Director New Market Initiatives Sayers Media Group VoIP Supply, LLC 454 Sonwil Drive Buffalo, NY 14225 716-250-3402 OFFICE 716-630-1548 FAX 716-601-4474 MOBILE candr...@sayersmedia.com Have I exceeded your expectations? Please share your experience with my boss, Benjamin P. Sayers, CEO NOTICE: The information contained in this email and any document attached hereto is intended only for the named recipient(s). It is the property of the VoIP Supply, LLC and shall not be used, disclosed or reproduced without the express written consent of VoIP Supply, LLC. If you are not the intended recipient, nor the employee or agent responsible for delivering this message in confidence to the intended recipient(s), you are hereby notified that you have received this transmittal in error, and any review, dissemination, distribution or copying of this transmittal or its attachments is strictly prohibited. If you have received this transmittal and/or attachments in error, please notify me immediately by reply e-mail or telephone and then delete this message, including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY 14225 USA. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Darrin Henshaw Sent: Tuesday, May 26, 2009 7:40 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Converting Cisco 7961 to SIP As part of a project to move a clients Cisco phones to SIP to work with an Avaya system, I'm taking a Cisco 7961 we bought and adding it to our Asterisk setup. Now, I've gotten the firmware files from the site, the latest version is 8.4 which contains the following files: apps41.8-4-3-16.sbn cnu41.8-4-3-16.sbn cvm41sip.8-4-3-16.sbn dsp41.8-4-3-16.sbn jar41sip.8-4-3-16.sbn SIP41.8-4-4S.loads term41.default.loads term61.default.loads Now I've read over loads of documentation on it, but am getting tripped up. Most of what I've seen talks about the older firmware versions usually 7.4. I have a feeling I'm still missing a lot of stuff. Anyone have any recent links or information? Also, anyone know of a decent way to generate the config files? I'd hate to have to go through all of it manually? Thanks. Cheers, Darrin Henshaw This email and its attachments may be confidential and are intended solely for the use of the individual or parties' to whom it is addressed. All comments are solely those of the author and do not necessarily represent those of Ignition. If you are not the intended recipient of this email and its attachments, you must take no action based upon them, nor must you copy or show them to anyone. Please contact the sender if you believe you have received this email in error. Thanks for considering the environmental impact before printing this email. ___ -- 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] Unable to make outbound calls
Based on this link - http://www.trixbox.org/forums/vendor-forums-certified/sangoma/a101dx-hangup- cause-code-90-outbound-calls I'd check my polarity settings in dahdi.conf. Maybe signaling? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kal Feher Sent: Tuesday, May 26, 2009 8:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Unable to make outbound calls Sorry. I don't get many opportunities to test this system as its live. Here are the results: -- Executing [...@dlpn_dialplan1:1] Dial(SIP/19722-b650fb80, DAHDI/1) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called 1 -- Channel 0/1, span 1 got hangup, cause 90 WARNING[6454]: app_dial.c:765 wait_for_answer: Unable to forward voice or dtmf -- Hungup 'DAHDI/1-1' == Everyone is busy/congested at this time (1:0/0/1) On 20/5/09 3:52 PM, Danny Nicholas da...@debsinc.com wrote: This all looks ok. What happens if you try to access the DAHDI channel outside of Asterisk control: In dialplan Exten = 9,1,Dial(DAHDI/1) Dial 9 Get dialtone Dial number -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kal Feher Sent: Wednesday, May 20, 2009 2:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Unable to make outbound calls I attached the show channels in my first post, but removed it to reduce the length of replies. Here it is again along with show status. Note that there is only 1 PRI currently attached. geriatrix*CLI dahdi show status Description Alarms IRQbpviol CRC4 T2XXP (PCI) Card 0 Span 1OK 0 0 0 T2XXP (PCI) Card 0 Span 2RED0 0 0 geriatrix*CLI dahdi show channels Chan Extension Context Language MOH Interpret pseudodefaultdefault 1DID_span_1 default 2DID_span_1 default 3DID_span_1 default 4DID_span_1 default 5DID_span_1 default 6DID_span_1 default 7DID_span_1 default 8DID_span_1 default 9DID_span_1 default 10DID_span_1 default 11DID_span_1 default 12DID_span_1 default 13DID_span_1 default 14DID_span_1 default 15DID_span_1 default 16DID_span_1 default 17DID_span_1 default 18DID_span_1 default 19DID_span_1 default 20DID_span_1 default 21DID_span_1 default 22DID_span_1 default 23DID_span_1 default 25DID_span_2 default 26DID_span_2 default 27DID_span_2 default 28DID_span_2 default 29DID_span_2 default 30DID_span_2 default 31DID_span_2 default 32DID_span_2 default 33DID_span_2 default 34DID_span_2 default 35DID_span_2 default 36DID_span_2 default 37DID_span_2 default 38DID_span_2 default 39DID_span_2 default 40DID_span_2 default 41DID_span_2 default 42DID_span_2 default 43DID_span_2 default 44DID_span_2 default 45DID_span_2 default 46DID_span_2 default 47DID_span_2 default On 19/5/09 6:31 PM, Danny Nicholas da...@debsinc.com wrote: Please post your CLI output from dahdi show status and dahdi show channels. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kal Feher Sent: Tuesday, May 19, 2009 11:22 AM To:
Re: [asterisk-users] Asterisk and Data Modem
Install nv_faxdetect. This will make asterisk not attempt to process the modem call for a specified period of time. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jon Morgan Sent: Tuesday, May 26, 2009 5:06 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Asterisk and Data Modem Hi All, We have an Asterisk 1.4.21.2 box which uses a 2 port Digium card to bridge calls, as follows: ISDN Provider --- Span 1(pri_cpe) --- Span 2(pri_net) Phone System The company that looks after our internal phone system can no longer dial in using their data modem in order to maintain the internal phone system. Is there any way we can configure our asterisk to allow them to dial in using their modem? Regards, Jon. ___ -- 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] 1.6.0.9 sip.c: Serious Network Trouble ??
On Saturday 23 May 2009 11:03:13 sean darcy wrote: I'm trying to upgrade from 1.4.24.1 to 1.6.0.9 over this weekend. I'm getting: [May 23 10:56:33] ERROR[26017]: chan_sip.c:2922 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data [May 23 10:56:33] ERROR[26017]: chan_sip.c:2922 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data [May 23 10:56:33] ERROR[26017]: chan_sip.c:2922 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data . What does this mean? What do i do about it? sip worked fine in 1.4.24.1. The difference is that 1.6.0 reports network errors, whereas 1.4 did not. -- 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] Unable to make outbound calls
My thoughts exactly. I've tried National2, 4ess and now ni1 ni1 just worked on Asterisk 1.4.22. (failover box I downgraded). So I'm swapping back to 1.4.24 to test that now. On 26/5/09 3:34 PM, Danny Nicholas da...@debsinc.com wrote: Based on this link - http://www.trixbox.org/forums/vendor-forums-certified/sangoma/a101dx-hangup- cause-code-90-outbound-calls I'd check my polarity settings in dahdi.conf. Maybe signaling? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kal Feher Sent: Tuesday, May 26, 2009 8:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Unable to make outbound calls Sorry. I don't get many opportunities to test this system as its live. Here are the results: -- Executing [...@dlpn_dialplan1:1] Dial(SIP/19722-b650fb80, DAHDI/1) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called 1 -- Channel 0/1, span 1 got hangup, cause 90 WARNING[6454]: app_dial.c:765 wait_for_answer: Unable to forward voice or dtmf -- Hungup 'DAHDI/1-1' == Everyone is busy/congested at this time (1:0/0/1) On 20/5/09 3:52 PM, Danny Nicholas da...@debsinc.com wrote: This all looks ok. What happens if you try to access the DAHDI channel outside of Asterisk control: In dialplan Exten = 9,1,Dial(DAHDI/1) Dial 9 Get dialtone Dial number -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kal Feher Sent: Wednesday, May 20, 2009 2:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Unable to make outbound calls I attached the show channels in my first post, but removed it to reduce the length of replies. Here it is again along with show status. Note that there is only 1 PRI currently attached. geriatrix*CLI dahdi show status Description Alarms IRQbpviol CRC4 T2XXP (PCI) Card 0 Span 1OK 0 0 0 T2XXP (PCI) Card 0 Span 2RED0 0 0 geriatrix*CLI dahdi show channels Chan Extension Context Language MOH Interpret pseudodefaultdefault 1DID_span_1 default 2DID_span_1 default 3DID_span_1 default 4DID_span_1 default 5DID_span_1 default 6DID_span_1 default 7DID_span_1 default 8DID_span_1 default 9DID_span_1 default 10DID_span_1 default 11DID_span_1 default 12DID_span_1 default 13DID_span_1 default 14DID_span_1 default 15DID_span_1 default 16DID_span_1 default 17DID_span_1 default 18DID_span_1 default 19DID_span_1 default 20DID_span_1 default 21DID_span_1 default 22DID_span_1 default 23DID_span_1 default 25DID_span_2 default 26DID_span_2 default 27DID_span_2 default 28DID_span_2 default 29DID_span_2 default 30DID_span_2 default 31DID_span_2 default 32DID_span_2 default 33DID_span_2 default 34DID_span_2 default 35DID_span_2 default 36DID_span_2 default 37DID_span_2 default 38DID_span_2 default 39DID_span_2 default 40DID_span_2 default 41DID_span_2 default 42DID_span_2 default 43DID_span_2 default 44DID_span_2 default 45DID_span_2 default 46DID_span_2 default 47DID_span_2 default On 19/5/09 6:31 PM, Danny Nicholas da...@debsinc.com wrote: Please post
Re: [asterisk-users] howto store local exchange prefixes ?
Now that I've slogged through everyone else's reply and got to the original post, here's an idea. You seem to have the dialplan part worked out; why not do a simple HTML interface to do the Berkley maint using asterisk -rx to do the CLI reads/pokes? With asterisk -rx you can automate 90+ percent of CLI functions. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy Sent: Monday, May 25, 2009 4:29 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] howto store local exchange prefixes ? The local telco is now going 10 digit dialing even for local (free) calls which used to be 7 digit. For a while no problem, everyone will continue to dial 7 digits, and I'll add the area code. But pretty soon everyone will become used to 10 digits. There are about 40 3 digit local exchanges. I'd like to store the exchanges in a database, and use the dialplan to check them. I can figure that out. I've looked at the Berkeley DB. That works pretty well, if the exchanges are all stored. But it looks like the exchanges have to be entered 1 by 1 from the CLI. And can only be reviewed, corrected, or deleted from the CLI. I haven't found any simple frontend for the DB. I'd also consider sqlite3, but from the sqlite3 .conf.sample, it's only for CDR. In any event, I couldn't find a simple frontend. I'd prefer not to go into mysql etc for such a simple project. 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 ___ -- 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] Problem running Dahdi
It is my experience that /e/i/dahdi doesn't always work correctly (opensuse 11.0). For whatever reason, it doesn't do the required modprobe to get the dadhi module activated. Try doing modprobe wctdm Then Dahdi_cfg -vv -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Monday, May 25, 2009 9:27 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Problem running Dahdi I did run make install, probably 3-4 times before I ended up asking that question in the mailing list. Here is the required output: to the first one, could not find module dahdi. To the second, it found dahdi in /lib/modules/2.6.18-128.1.10.el5/dahdi As for the other questions: What do you do? I simply try starting /etc/init.d/dahdi restart What would you expect to happen? No Red warnings, for one. I have another system that I configured awhile ago, and that starts fine. I understand I have no hardware loaded, but all modules load with a green OK. What actually happens? FATAL: Module Dahdi not found [snip] all modules listed as not found [/snip] Error: missing /dev/dahdi! What system is this on? - What versions of dahdi-linux and dahdi-tools? Latest as found on asterisk.org, that would be DAHDI Linux 2.1.0.4 DAHDI Tools 2.1.0.2 - What distribution? What version? CentOS, 5.3. I tried updating all packages before trying again, same result. - What kernel version? 2.6.18-128.1.10.el5 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Monday, May 25, 2009 9:53 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Problem running Dahdi On Mon, May 25, 2009 at 09:32:07AM -0400, Mike wrote: Sorry, it seems to have disappeared from my original email! FATAL: Module Dahdi not found [snip] all modules listed as not found [/snip] Error: missing /dev/dahdi! Your description makes me suspect you have not run 'make install' in dahdi-linux. What is the output of: modinfo dahdi find /lib/modules -name dahdi -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] Unable to make outbound calls
Ok I've solved the problem. I do not think it was as switchtype issue after all as it is now working with a national2 configuration. I need to sort out some of the changes and I'll post back for reference. However it appears to be some form of parsing order issue between all the locations that define dahdi trunk groups. What is odd is that this appears anecdotally to be different between 1.4.22 and 1.4.24. But I'll confirm and reply. On 26/5/09 3:46 PM, Kal Feher kalman.fe...@melbourneit.com.au wrote: My thoughts exactly. I've tried National2, 4ess and now ni1 ni1 just worked on Asterisk 1.4.22. (failover box I downgraded). So I'm swapping back to 1.4.24 to test that now. On 26/5/09 3:34 PM, Danny Nicholas da...@debsinc.com wrote: Based on this link - http://www.trixbox.org/forums/vendor-forums-certified/sangoma/a101dx-hangup- cause-code-90-outbound-calls I'd check my polarity settings in dahdi.conf. Maybe signaling? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kal Feher Sent: Tuesday, May 26, 2009 8:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Unable to make outbound calls Sorry. I don't get many opportunities to test this system as its live. Here are the results: -- Executing [...@dlpn_dialplan1:1] Dial(SIP/19722-b650fb80, DAHDI/1) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called 1 -- Channel 0/1, span 1 got hangup, cause 90 WARNING[6454]: app_dial.c:765 wait_for_answer: Unable to forward voice or dtmf -- Hungup 'DAHDI/1-1' == Everyone is busy/congested at this time (1:0/0/1) On 20/5/09 3:52 PM, Danny Nicholas da...@debsinc.com wrote: This all looks ok. What happens if you try to access the DAHDI channel outside of Asterisk control: In dialplan Exten = 9,1,Dial(DAHDI/1) Dial 9 Get dialtone Dial number -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kal Feher Sent: Wednesday, May 20, 2009 2:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Unable to make outbound calls I attached the show channels in my first post, but removed it to reduce the length of replies. Here it is again along with show status. Note that there is only 1 PRI currently attached. geriatrix*CLI dahdi show status Description Alarms IRQbpviol CRC4 T2XXP (PCI) Card 0 Span 1OK 0 0 0 T2XXP (PCI) Card 0 Span 2RED0 0 0 geriatrix*CLI dahdi show channels Chan Extension Context Language MOH Interpret pseudodefaultdefault 1DID_span_1 default 2DID_span_1 default 3DID_span_1 default 4DID_span_1 default 5DID_span_1 default 6DID_span_1 default 7DID_span_1 default 8DID_span_1 default 9DID_span_1 default 10DID_span_1 default 11DID_span_1 default 12DID_span_1 default 13DID_span_1 default 14DID_span_1 default 15DID_span_1 default 16DID_span_1 default 17DID_span_1 default 18DID_span_1 default 19DID_span_1 default 20DID_span_1 default 21DID_span_1 default 22DID_span_1 default 23DID_span_1 default 25DID_span_2 default 26DID_span_2 default 27DID_span_2 default 28DID_span_2 default 29DID_span_2 default 30DID_span_2 default 31DID_span_2 default 32DID_span_2 default 33DID_span_2 default 34DID_span_2 default 35DID_span_2 default 36DID_span_2 default 37DID_span_2 default 38DID_span_2 default 39DID_span_2 default
Re: [asterisk-users] Converting Cisco 7961 to SIP
Ok, ignore what I said below. I've got it working now, thanks a million for this link: http://www.greenwireit.com/blog/2009/04/reflash-your-cisco-7940-7941-7960-or-7961-phone-to-sip/. However, now I'm wondering about the dialplan.xml, can it handle regular expressions like 9[2-9]..? Thanks. Cheers, Darrin Henshaw | IT Administrator | MCTS: Exchange 2007 | MCSE 2003 | LPIC Ignition Support Center | www.ignition.bm Bermuda (441) 496-4319 | Cayman (345) 947-4357 | Halifax (902) 482-1288 Atlanta | Bermuda | Cayman | Halifax -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Darrin Henshaw Sent: Tuesday, May 26, 2009 08:40 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Converting Cisco 7961 to SIP As part of a project to move a clients Cisco phones to SIP to work with an Avaya system, I'm taking a Cisco 7961 we bought and adding it to our Asterisk setup. Now, I've gotten the firmware files from the site, the latest version is 8.4 which contains the following files: apps41.8-4-3-16.sbn cnu41.8-4-3-16.sbn cvm41sip.8-4-3-16.sbn dsp41.8-4-3-16.sbn jar41sip.8-4-3-16.sbn SIP41.8-4-4S.loads term41.default.loads term61.default.loads Now I've read over loads of documentation on it, but am getting tripped up. Most of what I've seen talks about the older firmware versions usually 7.4. I have a feeling I'm still missing a lot of stuff. Anyone have any recent links or information? Also, anyone know of a decent way to generate the config files? I'd hate to have to go through all of it manually? Thanks. Cheers, Darrin Henshaw This email and its attachments may be confidential and are intended solely for the use of the individual or parties' to whom it is addressed. All comments are solely those of the author and do not necessarily represent those of Ignition. If you are not the intended recipient of this email and its attachments, you must take no action based upon them, nor must you copy or show them to anyone. Please contact the sender if you believe you have received this email in error. Thanks for considering the environmental impact before printing this email. ___ -- 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 This email and its attachments may be confidential and are intended solely for the use of the individual or parties' to whom it is addressed. All comments are solely those of the author and do not necessarily represent those of Ignition. If you are not the intended recipient of this email and its attachments, you must take no action based upon them, nor must you copy or show them to anyone. Please contact the sender if you believe you have received this email in error. Thanks for considering the environmental impact before printing this email. ___ -- 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] A problem in playing sound files
Hello, I have 8 DID: 7 from a provider1 and 1 from provider2. Each time a customer calls one of the DID, the system plays a message. The problem is that the message is played normally for all the DIDs from the provider1 and is not played (not heard) for the DID from provider2. My question is: What can be the cause of this problem. Thanks. ___ -- 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] Maximum cable length for analog phone from FXS port
Hello. I am looking for details of the maximum allowed/usable/effective wire/cable length of the connection from a FXS port of Digium analog cards to the analog telephone handset. To clarify my intention, I need to have an analog telephone connection to my asterisk box that is 3000 meters (3km) away at least. If you have any details of ATA boxes or other similar devices that I could use to do this, I'd appreciate your input. It must be able to use a regular analog telephone handset on the far end. I've searched high and low and either I'm not clever enough in using the right terms for this or it is rarely documented? Any details much appreciated. Thank you! Baldvin ___ -- 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] Problem running Dahdi
Thanks for taking the time to answer. I've played with the server a lot in the past few days, and I am not sure what did it, but for futur reference this is my best guess: I think I had 32-bit code or RPMs installed on a 64-bit machine (specifically: HP-hardware specific RPMs for hardware monitoring). Things seemed well on the surface, but weren't going too great under the hood. Fixed now. Thanks for those who took the time to try and help. Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Tuesday, May 26, 2009 10:04 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Problem running Dahdi It is my experience that /e/i/dahdi doesn't always work correctly (opensuse 11.0). For whatever reason, it doesn't do the required modprobe to get the dadhi module activated. Try doing modprobe wctdm Then Dahdi_cfg -vv -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Monday, May 25, 2009 9:27 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Problem running Dahdi I did run make install, probably 3-4 times before I ended up asking that question in the mailing list. Here is the required output: to the first one, could not find module dahdi. To the second, it found dahdi in /lib/modules/2.6.18-128.1.10.el5/dahdi As for the other questions: What do you do? I simply try starting /etc/init.d/dahdi restart What would you expect to happen? No Red warnings, for one. I have another system that I configured awhile ago, and that starts fine. I understand I have no hardware loaded, but all modules load with a green OK. What actually happens? FATAL: Module Dahdi not found [snip] all modules listed as not found [/snip] Error: missing /dev/dahdi! What system is this on? - What versions of dahdi-linux and dahdi-tools? Latest as found on asterisk.org, that would be DAHDI Linux 2.1.0.4 DAHDI Tools 2.1.0.2 - What distribution? What version? CentOS, 5.3. I tried updating all packages before trying again, same result. - What kernel version? 2.6.18-128.1.10.el5 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Monday, May 25, 2009 9:53 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Problem running Dahdi On Mon, May 25, 2009 at 09:32:07AM -0400, Mike wrote: Sorry, it seems to have disappeared from my original email! FATAL: Module Dahdi not found [snip] all modules listed as not found [/snip] Error: missing /dev/dahdi! Your description makes me suspect you have not run 'make install' in dahdi-linux. What is the output of: modinfo dahdi find /lib/modules -name dahdi -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] h extension and channel variables
On 5/26/2009 14:08, Marco Sambo wrote:
I set a variable CalledID to ${EXTEN} before dial it. So in h extension
I can use ${CalledID}.
Thanks for the response.
In that case if there is an intervening call that is shorter, then the
$calledID will be wrong.
I found a better approach than using the h, extensions.
___
-- 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] Converting Cisco 7961 to SIP
Ahh I see. In response to your other question about the auto-provisioning of Cisco phones, I wrote some scripts that work against an active directory and setup the phones automagically. I'll send the link your way if you'd like. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cory Andrews Sent: Tuesday, May 26, 2009 10:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Converting Cisco 7961 to SIP Did not mean to infer they don't perform wonderfully with Asterisk. By hack I meant that Cisco does not offer any official support for them on Asterisk. Cory J. Andrews Director New Market Initiatives Sayers Media Group VoIP Supply, LLC 454 Sonwil Drive Buffalo, NY 14225 716-250-3402 OFFICE 716-630-1548 FAX 716-601-4474 MOBILE candr...@sayersmedia.com Have I exceeded your expectations? Please share your experience with my boss, Benjamin P. Sayers, CEO NOTICE: The information contained in this email and any document attached hereto is intended only for the named recipient(s). It is the property of the VoIP Supply, LLC and shall not be used, disclosed or reproduced without the express written consent of VoIP Supply, LLC. If you are not the intended recipient, nor the employee or agent responsible for delivering this message in confidence to the intended recipient(s), you are hereby notified that you have received this transmittal in error, and any review, dissemination, distribution or copying of this transmittal or its attachments is strictly prohibited. If you have received this transmittal and/or attachments in error, please notify me immediately by reply e-mail or telephone and then delete this message, including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY 14225 USA. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: Tuesday, May 26, 2009 10:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Converting Cisco 7961 to SIP Cory, Precisely what do you mean by 'Anything other than Callmanager will essentially be a hack'? I've got nearly 100 Cisco 79x1s in production using Asterisk against the SIP image. They're not 'hacked', they're set up properly against the Cisco provided SIP image and are rock-solid stable. I would pit them against any of the cheaper model SIP phones any time, any place, any day. I've written scripts to do nearly everything that call manager can do without paying hundreds of dollars per user for the call manager software. Just about the only thing they can't do at the moment is BLF because they require SIP over TCP to handle SIP messages about BLF status, something that I'm not willing to implement just yet. In the past, Cisco phones have had a bad rap as not being usable outside of a call manager environment. That's just not the case. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cory Andrews Sent: Tuesday, May 26, 2009 9:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Converting Cisco 7961 to SIP Darrin, The files you are using are consistent with SIP for Cisco Call Manager. Anything other than Callmanager will essentially be a hack. I am not sure how proprietary the Avaya system is in regards to registration and open-SIP support. Asterisk and any iteration of it will support it, but Cisco hasn't really designed a load compatible with it yet. I can tell you that I haven't really found any configuration file generation tools for these files. The reason being is that these loads are mainly used for SCCP and SIP Cisco systems. There is a well known tutorial on how to Hack to the CP-7970 to trixbox CE located here: http://www.asterisktutorials.com/cisco-7970-ip-phone/ This may help get you jump started and pointed in the right direction. The only problem that may arise is that in the tutorial, the use a specific SIP load (8.0.3) which may not be available for the 7961G. Cory J. Andrews Director New Market Initiatives Sayers Media Group VoIP Supply, LLC 454 Sonwil Drive Buffalo, NY 14225 716-250-3402 OFFICE 716-630-1548 FAX 716-601-4474 MOBILE candr...@sayersmedia.com Have I exceeded your expectations? Please share your experience with my boss, Benjamin P. Sayers, CEO NOTICE: The information contained in this email and any document attached hereto is intended only for the named recipient(s). It is the property of the VoIP Supply, LLC and shall not be used, disclosed or reproduced without the express written consent of VoIP Supply, LLC. If you are not the intended recipient, nor the employee or agent responsible for delivering this message in confidence to the intended
Re: [asterisk-users] Maximum cable length for analog phone from FXS port
The best a native cat5 can run is 100 meters. Unless you like paying your telco huge bucks, you should go for some kind of SIP connection to your box. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk-us...@rogg.is Sent: Tuesday, May 26, 2009 9:09 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Maximum cable length for analog phone from FXS port Hello. I am looking for details of the maximum allowed/usable/effective wire/cable length of the connection from a FXS port of Digium analog cards to the analog telephone handset. To clarify my intention, I need to have an analog telephone connection to my asterisk box that is 3000 meters (3km) away at least. If you have any details of ATA boxes or other similar devices that I could use to do this, I'd appreciate your input. It must be able to use a regular analog telephone handset on the far end. I've searched high and low and either I'm not clever enough in using the right terms for this or it is rarely documented? Any details much appreciated. Thank you! Baldvin ___ -- 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] Converting Cisco 7961 to SIP
Did not mean to infer they don't perform wonderfully with Asterisk. By hack I meant that Cisco does not offer any official support for them on Asterisk. Cory J. Andrews Director New Market Initiatives Sayers Media Group VoIP Supply, LLC 454 Sonwil Drive Buffalo, NY 14225 716-250-3402 OFFICE 716-630-1548 FAX 716-601-4474 MOBILE candr...@sayersmedia.com Have I exceeded your expectations? Please share your experience with my boss, Benjamin P. Sayers, CEO NOTICE: The information contained in this email and any document attached hereto is intended only for the named recipient(s). It is the property of the VoIP Supply, LLC and shall not be used, disclosed or reproduced without the express written consent of VoIP Supply, LLC. If you are not the intended recipient, nor the employee or agent responsible for delivering this message in confidence to the intended recipient(s), you are hereby notified that you have received this transmittal in error, and any review, dissemination, distribution or copying of this transmittal or its attachments is strictly prohibited. If you have received this transmittal and/or attachments in error, please notify me immediately by reply e-mail or telephone and then delete this message, including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY 14225 USA. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: Tuesday, May 26, 2009 10:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Converting Cisco 7961 to SIP Cory, Precisely what do you mean by 'Anything other than Callmanager will essentially be a hack'? I've got nearly 100 Cisco 79x1s in production using Asterisk against the SIP image. They're not 'hacked', they're set up properly against the Cisco provided SIP image and are rock-solid stable. I would pit them against any of the cheaper model SIP phones any time, any place, any day. I've written scripts to do nearly everything that call manager can do without paying hundreds of dollars per user for the call manager software. Just about the only thing they can't do at the moment is BLF because they require SIP over TCP to handle SIP messages about BLF status, something that I'm not willing to implement just yet. In the past, Cisco phones have had a bad rap as not being usable outside of a call manager environment. That's just not the case. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cory Andrews Sent: Tuesday, May 26, 2009 9:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Converting Cisco 7961 to SIP Darrin, The files you are using are consistent with SIP for Cisco Call Manager. Anything other than Callmanager will essentially be a hack. I am not sure how proprietary the Avaya system is in regards to registration and open-SIP support. Asterisk and any iteration of it will support it, but Cisco hasn't really designed a load compatible with it yet. I can tell you that I haven't really found any configuration file generation tools for these files. The reason being is that these loads are mainly used for SCCP and SIP Cisco systems. There is a well known tutorial on how to Hack to the CP-7970 to trixbox CE located here: http://www.asterisktutorials.com/cisco-7970-ip-phone/ This may help get you jump started and pointed in the right direction. The only problem that may arise is that in the tutorial, the use a specific SIP load (8.0.3) which may not be available for the 7961G. Cory J. Andrews Director New Market Initiatives Sayers Media Group VoIP Supply, LLC 454 Sonwil Drive Buffalo, NY 14225 716-250-3402 OFFICE 716-630-1548 FAX 716-601-4474 MOBILE candr...@sayersmedia.com Have I exceeded your expectations? Please share your experience with my boss, Benjamin P. Sayers, CEO NOTICE: The information contained in this email and any document attached hereto is intended only for the named recipient(s). It is the property of the VoIP Supply, LLC and shall not be used, disclosed or reproduced without the express written consent of VoIP Supply, LLC. If you are not the intended recipient, nor the employee or agent responsible for delivering this message in confidence to the intended recipient(s), you are hereby notified that you have received this transmittal in error, and any review, dissemination, distribution or copying of this transmittal or its attachments is strictly prohibited. If you have received this transmittal and/or attachments in error, please notify me immediately by reply e-mail or telephone and then delete this message, including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY 14225 USA. -Original Message- From: asterisk-users-boun...@lists.digium.com
Re: [asterisk-users] Maximum cable length for analog phone from FXS port
asterisk-us...@rogg.is wrote: Hello. I am looking for details of the maximum allowed/usable/effective wire/cable length of the connection from a FXS port of Digium analog cards to the analog telephone handset. To clarify my intention, I need to have an analog telephone connection to my asterisk box that is 3000 meters (3km) away at least. If you have any details of ATA boxes or other similar devices that I could use to do this, I'd appreciate your input. It must be able to use a regular analog telephone handset on the far end. I've searched high and low and either I'm not clever enough in using the right terms for this or it is rarely documented? Any details much appreciated. Thank you! Baldvin It's not expressed in distance. They will supply the current voltage output and you need to apply ohm's law. That requires knowing the resistance of the cable which is dependent on length and gauge. Lyle Giese LCR Computer Services, Inc. ___ -- 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] Maximum cable length for analog phone from FXS port
I could be wrong but I don't think the cat5 limit of 100 meters applies to any analog signaling over that copper. I believe it only applies to Ethernet signaling. -Dave From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Tuesday, May 26, 2009 10:41 AM To: bald...@rogg.is; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Maximum cable length for analog phone from FXS port The best a native cat5 can run is 100 meters. Unless you like paying your telco huge bucks, you should go for some kind of SIP connection to your box. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk-us...@rogg.is Sent: Tuesday, May 26, 2009 9:09 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Maximum cable length for analog phone from FXS port Hello. I am looking for details of the maximum allowed/usable/effective wire/cable length of the connection from a FXS port of Digium analog cards to the analog telephone handset. To clarify my intention, I need to have an analog telephone connection to my asterisk box that is 3000 meters (3km) away at least. If you have any details of ATA boxes or other similar devices that I could use to do this, I'd appreciate your input. It must be able to use a regular analog telephone handset on the far end. I've searched high and low and either I'm not clever enough in using the right terms for this or it is rarely documented? Any details much appreciated. Thank you! Baldvin ___ -- 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] Problem running Dahdi
On Mon, May 25, 2009 at 10:27:22AM -0400, Mike wrote: I did run make install, probably 3-4 times before I ended up asking that question in the mailing list. Here is the required output: to the first one, could not find module dahdi. To the second, it found dahdi in /lib/modules/2.6.18-128.1.10.el5/dahdi Either you have not run depmod (which is strange, it is part of 'make install') or '2.6.18-128.1.10.el5' is not your kernel version. - What kernel version? 2.6.18-128.1.10.el5 What is the output of: uname -r If it is exactly '2.6.18-128.1.10.el5' , then try: depmod modinfo dahdi -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-addon 1.6.1 problem
On Tuesday 26 May 2009 02:52:18 Rilawich Ango wrote: Hi all, I download asterisk-addon 1.6.1 but the VoIP phone failed to register to the system with the message below. [May 26 15:45:11] WARNING[29665]: res_config_mysql.c:317 realtime_mysql: MySQL RealTime: Invalid database specified: asterisk [May 26 15:45:11] WARNING[29665]: res_config_mysql.c:317 realtime_mysql: MySQL RealTime: Invalid database specified: asterisk sip I use the same configuration file (res_mysql.conf extconfig.conf) in 1.6.0 but failed. Any big change in 1.6.1? Please read UPGRADE.txt in the asterisk-addons directory. -- 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] FXS
There is indeed... well i was about to say there was, but it turns out the one i've got is an fxo adapter, have a look and see if sangoma have any fxs adapters in the series, it seems to be called the usbfxo u100 2009/5/26 Diogo Saad diogos...@gmail.com What is the easiest way to connect my black phone to a PC running asterisk? I don't need multiple extensions, I've got just 1 phone. Is there any USB FXS adapter? Thanks -- Diogo Saad ___ -- 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] FXS
what I want to do is to answers to mobile calls using a regular phone. Is a usb fxs all I need? Does this u100 have smooth integration with Asterisk ? On Tue, May 26, 2009 at 11:55 AM, Geraint Lee gera...@gmail.com wrote: There is indeed... well i was about to say there was, but it turns out the one i've got is an fxo adapter, have a look and see if sangoma have any fxs adapters in the series, it seems to be called the usbfxo u100 2009/5/26 Diogo Saad diogos...@gmail.com What is the easiest way to connect my black phone to a PC running asterisk? I don't need multiple extensions, I've got just 1 phone. Is there any USB FXS adapter? Thanks -- Diogo Saad ___ -- 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 -- Diogo Saad ___ -- 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] Maximum cable length for analog phone from FXS port
On Tue, May 26, 2009 at 5:31 PM, Danny Nicholas da...@debsinc.com wrote: I run my analog telco over cat5, but that's in-house and definitely not 3km. That sounds really far for current loop stuff. I was doing that too. I asked this same question a few years ago and the answer was 100-200 meters. This is just a quick rule of thumb, but it seems about right. 3km, I doubt that would work, but it depends, as someone said, totally depending on ohm's law :) r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maximum cable length for analog phone from FXS port
On Tue, 26 May 2009, Steve Howes wrote: On 26 May 2009, at 16:39, Jeff LaCoursiere wrote: YMMV I think thats the problem :D sorry couldn't resist.. I did kind of mean that tounge-in-cheek :):) j ___ -- 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] Maximum cable length for analog phone from FXS port
On 26 May 2009, at 16:39, Jeff LaCoursiere wrote: YMMV I think thats the problem :D sorry couldn't resist.. ___ -- 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] Maximum cable length for analog phone from FXS port
On Tue, 26 May 2009, randulo wrote: On Tue, May 26, 2009 at 5:31 PM, Danny Nicholas da...@debsinc.com wrote: I run my analog telco over cat5, but that's in-house and definitely not 3km. That sounds really far for current loop stuff. I was doing that too. I asked this same question a few years ago and the answer was 100-200 meters. This is just a quick rule of thumb, but it seems about right. 3km, I doubt that would work, but it depends, as someone said, totally depending on ohm's law :) Egad, this is just not true. 100 - 200 meters is for ETHERNET, not analog voice. I have many runs over 1K meters that work just fine, and several that are close to 3Km that honestly do NOT. Think about high rise buildings - many strung with CAT3 cable for voice from the basement. Many of those runs may be well over 1000m. A better question is why is he stuck using a 3Km leased circuit? Like another poster said there must be a better way. j ___ -- 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] Maximum cable length for analog phone from FXS port
That is a pretty long run. The type of analog phone can be an issue. How LITTLE loop current will it operate on? Most need more than 20 Ma to signal properly, and the voltage output of the ATA needs to be known Type of signaling? DTMF? pulse? Interconnection cable wire size and capacitance will affect high frequency response, loop current, inductive pickup and pulse shaping to name just a few. The ATA requirements need to be known. A total loop resistance of 500 ohms should work, but go out to 1200 and most will fail Do you really have control over this or will you be renting a pair from the local telco? Protection should be applied on both ends for safety of the user(s) and devices. There MUST be a better way??? asterisk-us...@rogg.is wrote: Hello. I am looking for details of the maximum allowed/usable/effective wire/cable length of the connection from a FXS port of Digium analog cards to the analog telephone handset. To clarify my intention, I need to have an analog telephone connection to my asterisk box that is 3000 meters (3km) away at least. If you have any details of ATA boxes or other similar devices that I could use to do this, I'd appreciate your input. It must be able to use a regular analog telephone handset on the far end. I've searched high and low and either I'm not clever enough in using the right terms for this or it is rarely documented? Any details much appreciated. Thank you! Baldvin ___ -- 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 -- Dog is my co-pilot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] howto store local exchange prefixes ?
sean darcy wrote:
Maybe I've not explained this correctly. I know, or can look up, the 40+
local exchanges that are local. I can parse the dial EXTEN to determine
the exchange. I can check the exchange against a DB. I want to determine
which exchanges are local. I do not want to store an exchange dialed
by a user.
I didn't explain myself very well.
My Asterisk system sits between the PSTN and a legacy PBX. Asterisk
answers the call and among other things, prompts for an extension
number. I needed to know if the extension entered is valid before
sending the call on to the old PBX. I simply have a lookup subroutine
to validate the extension.
My code for looking up the validity of their entry is:
exten = _[123]XX,1,Verbose(1,${CALLERID(all)} requested extension
${EXTEN});
exten = _[123]XX,n,Gosub(validate-extension,s,1(${EXTEN}));
exten = _[123]XX,n,Goto(extension-${GOSUB_RETVAL});
exten = _[123]XX,n(extension-FOUND),Verbose(1,${CALLERID(all)} xfer to
${DB(${DB_IWATSU_EXTENSIONS}/${EXTEN})} at extension ${EXTEN});
exten = _[123]XX,n,macro(bridge-to-iwatsu,7${EXTEN});
exten = _[123]XX,n(extension-NOTFOUND),background(pbx-invalid);
exten = _[123]XX,n,WaitExten(5);
The lookup, which will initialize the AsteriskDB if necessary, is:
;
; This subroutine's purpose is to check the validity of an extension.
;
; Parameters:
; ARG1 = Extension to check
; Returns:
; FOUND or NOTFOUND
;
[validate-extension]
exten = s,1,Verbose(1,Checking validity of extension ${ARG1});
;
; Let's check to ensure that the database is loaded. We'll do
; that by looking for extension 399, the Iwatsu master phone.
;
exten = s,n,GotoIf(${DB_EXISTS(${DB_IWATSU_EXTENSIONS}/399)}?search:load)
exten = s,n(load),DBdeltree(${DB_IWATSU_EXTENSIONS}); Clear all
existing entries
exten = s,n,Set(DB(${DB_IWATSU_EXTENSIONS}/120)='Rikki')
exten = s,n,Set(DB(${DB_IWATSU_EXTENSIONS}/121)='Terri')
exten = s,n,Set(DB(${DB_IWATSU_EXTENSIONS}/122)='CorpConf')
exten = s,n,Set(DB(${DB_IWATSU_EXTENSIONS}/123)='Linda')
exten = s,n,Set(DB(${DB_IWATSU_EXTENSIONS}/124)='Kim')
exten = s,n,Set(DB(${DB_IWATSU_EXTENSIONS}/125)='Nancy B')
exten = s,n,Set(DB(${DB_IWATSU_EXTENSIONS}/126)='Wayne')
...
;
; Extension 399 is the master extension for the Iwatsu
; and should always show up. It is used for testing
; the validity of the database in the dialplan.
;
exten = s,n,Set(DB(${DB_IWATSU_EXTENSIONS}/399)='MASTER')
;
; Search here
;
exten =
s,n(search),GotoIf(${DB_EXISTS(${DB_IWATSU_EXTENSIONS}/${ARG1})}?found:notfound)
exten = s,n(found),Return(FOUND);
exten = s,n(notfound),Return(NOTFOUND);
___
-- 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] Converting Cisco 7961 to SIP
Cory, Precisely what do you mean by 'Anything other than Callmanager will essentially be a hack'? I've got nearly 100 Cisco 79x1s in production using Asterisk against the SIP image. They're not 'hacked', they're set up properly against the Cisco provided SIP image and are rock-solid stable. I would pit them against any of the cheaper model SIP phones any time, any place, any day. I've written scripts to do nearly everything that call manager can do without paying hundreds of dollars per user for the call manager software. Just about the only thing they can't do at the moment is BLF because they require SIP over TCP to handle SIP messages about BLF status, something that I'm not willing to implement just yet. In the past, Cisco phones have had a bad rap as not being usable outside of a call manager environment. That's just not the case. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cory Andrews Sent: Tuesday, May 26, 2009 9:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Converting Cisco 7961 to SIP Darrin, The files you are using are consistent with SIP for Cisco Call Manager. Anything other than Callmanager will essentially be a hack. I am not sure how proprietary the Avaya system is in regards to registration and open-SIP support. Asterisk and any iteration of it will support it, but Cisco hasn't really designed a load compatible with it yet. I can tell you that I haven't really found any configuration file generation tools for these files. The reason being is that these loads are mainly used for SCCP and SIP Cisco systems. There is a well known tutorial on how to Hack to the CP-7970 to trixbox CE located here: http://www.asterisktutorials.com/cisco-7970-ip-phone/ This may help get you jump started and pointed in the right direction. The only problem that may arise is that in the tutorial, the use a specific SIP load (8.0.3) which may not be available for the 7961G. Cory J. Andrews Director New Market Initiatives Sayers Media Group VoIP Supply, LLC 454 Sonwil Drive Buffalo, NY 14225 716-250-3402 OFFICE 716-630-1548 FAX 716-601-4474 MOBILE candr...@sayersmedia.com Have I exceeded your expectations? Please share your experience with my boss, Benjamin P. Sayers, CEO NOTICE: The information contained in this email and any document attached hereto is intended only for the named recipient(s). It is the property of the VoIP Supply, LLC and shall not be used, disclosed or reproduced without the express written consent of VoIP Supply, LLC. If you are not the intended recipient, nor the employee or agent responsible for delivering this message in confidence to the intended recipient(s), you are hereby notified that you have received this transmittal in error, and any review, dissemination, distribution or copying of this transmittal or its attachments is strictly prohibited. If you have received this transmittal and/or attachments in error, please notify me immediately by reply e-mail or telephone and then delete this message, including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY 14225 USA. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Darrin Henshaw Sent: Tuesday, May 26, 2009 7:40 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Converting Cisco 7961 to SIP As part of a project to move a clients Cisco phones to SIP to work with an Avaya system, I'm taking a Cisco 7961 we bought and adding it to our Asterisk setup. Now, I've gotten the firmware files from the site, the latest version is 8.4 which contains the following files: apps41.8-4-3-16.sbn cnu41.8-4-3-16.sbn cvm41sip.8-4-3-16.sbn dsp41.8-4-3-16.sbn jar41sip.8-4-3-16.sbn SIP41.8-4-4S.loads term41.default.loads term61.default.loads Now I've read over loads of documentation on it, but am getting tripped up. Most of what I've seen talks about the older firmware versions usually 7.4. I have a feeling I'm still missing a lot of stuff. Anyone have any recent links or information? Also, anyone know of a decent way to generate the config files? I'd hate to have to go through all of it manually? Thanks. Cheers, Darrin Henshaw This email and its attachments may be confidential and are intended solely for the use of the individual or parties' to whom it is addressed. All comments are solely those of the author and do not necessarily represent those of Ignition. If you are not the intended recipient of this email and its attachments, you must take no action based upon them, nor must you copy or show them to anyone. Please contact the sender if you believe you have received this email in error. Thanks for considering the environmental impact before printing this email. ___ --
Re: [asterisk-users] Maximum cable length for analog phone from FXS port
On Tue, May 26, 2009 at 05:39:46PM +0200, randulo wrote: On Tue, May 26, 2009 at 5:31 PM, Danny Nicholas da...@debsinc.com wrote: I run my analog telco over cat5, but that's in-house and definitely not 3km. That sounds really far for current loop stuff. I was doing that too. I asked this same question a few years ago and the answer was 100-200 meters. This is just a quick rule of thumb, but it seems about right. 3km, I doubt that would work, but it depends, as someone said, totally depending on ohm's law :) If it were to depend solely on Ohm's law, than 3km would be marginal but probably within reach. Not exactly sure what other factors are there to count. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS
On Tue, 26 May 2009, Diogo Saad wrote: What is the easiest way to connect my black phone to a PC running asterisk? I don't need multiple extensions, I've got just 1 phone. Is there any USB FXS adapter? An Ethernet based ATA would be more versatile. I like Digium's discontinued IAXy. Dead simple to configure, easy to travel with, no NAT headaches. Used on ebay should set you back about US$30. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maximum cable length for analog phone from FXSport
Sigh, lets repeal Ohm's law. ;-) In practice the controlling rules are: Murphy's Law: If anything can go wrong it will. O'Toole's corollary to Murphy's law: And, it will produce the worst possible results. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Tuesday, May 26, 2009 10:56 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Maximum cable length for analog phone from FXSport On Tue, May 26, 2009 at 05:39:46PM +0200, randulo wrote: On Tue, May 26, 2009 at 5:31 PM, Danny Nicholas da...@debsinc.com wrote: I run my analog telco over cat5, but that's in-house and definitely not 3km. That sounds really far for current loop stuff. I was doing that too. I asked this same question a few years ago and the answer was 100-200 meters. This is just a quick rule of thumb, but it seems about right. 3km, I doubt that would work, but it depends, as someone said, totally depending on ohm's law :) If it were to depend solely on Ohm's law, than 3km would be marginal but probably within reach. Not exactly sure what other factors are there to count. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FXS
What is the easiest way to connect my black phone to a PC running asterisk? I don't need multiple extensions, I've got just 1 phone. Is there any USB FXS adapter? Thanks -- Diogo Saad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to register with TCP transport ?
Hi, In my sip.conf, I've got : [general](+) ; register=tcp://trunk4ipbx:passw...@192.168.100.129trunk4ipbx%3apassw...@192.168.100.129 register=trunk4ipbx:passw...@192.168.100.129trunk4ipbx%3apassw...@192.168.100.129 When I'm using the TCP line instead of the other, I've got : [May 26 17:58:42] NOTICE[2859]: chan_sip.c:20169 sip_parse_host: '/' is not a valid port number on line 25 of sip.conf. using default. [May 26 17:58:42] WARNING[2859]: chan_sip.c:6560 sip_register: Format for registration is [transport://]user[:secret[:authuse...@domain[:port][/extension][~expiry] at line 25 Is this register=tcp://trunk4ipbx:passw...@192.168.100.129trunk4ipbx%3apassw...@192.168.100.129 statement correct ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maximum cable length for analog phone from FXS port
randulo escribió: On Tue, May 26, 2009 at 5:31 PM, Danny Nicholas da...@debsinc.com wrote: I run my analog telco over cat5, but that's in-house and definitely not 3km. That sounds really far for current loop stuff. I was doing that too. I asked this same question a few years ago and the answer was 100-200 meters. This is just a quick rule of thumb, but it seems about right. 3km, I doubt that would work, but it depends, as someone said, totally depending on ohm's law :) What I think about this is, the length of the copper cable between the central office and home is usually several km, but definitely helped by the central office circuitry (current source instead of voltage source, that guarantees a minimum ringing voltage on the far end). What I don't know is, a FXS port behaves the same as a central office, electrically speaking? If that is so, you could extend your 3km of cable without problems, but I think you can have some noise problems depending on what places the cable has to go through. r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57 1)6500816 Móvil: (+57)3138873587 ___ -- 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] Maximum cable length for analog phone from FXS port
On Tue, May 26, 2009 at 10:09 AM, asterisk-us...@rogg.is wrote: I am looking for details of the maximum allowed/usable/effective wire/cable length of the connection from a FXS port of Digium analog cards to the analog telephone handset. To clarify my intention, I need to have an analog telephone connection to my asterisk box that is 3000 meters (3km) away at least. If you have any details of ATA boxes or other similar devices that I could use to do this, I‘d appreciate your input. It must be able to use a regular analog telephone handset on the far end. I‘ve searched high and low and either I‘m not clever enough in using the right terms for this or it is rarely documented? I've not seen a high-current ATA, but you could probably add a KIT8L from here: http://www.sandman.com/longloop.html to a regular terminal adapter to boost the loop voltage/current. -- Heath Roberts htrobe...@gmail.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maximum cable length for analog phone from FXS port
Also be wary of the loop you get. Depending on the Telco you are dealing with, and the type of loop you get, Alarm circuit, etc. they may , and have the right to, put in a low pass circuit to limit bandwidth to 15 Hz. That keeps people from using cheap alarm circuits for voice. It is not likely they will go to the trouble. But, they can do it. Cary Fitch _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Novack Sent: Tuesday, May 26, 2009 9:26 AM To: bald...@rogg.is; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Maximum cable length for analog phone from FXS port That is a pretty long run. The type of analog phone can be an issue. How LITTLE loop current will it operate on? Most need more than 20 Ma to signal properly, and the voltage output of the ATA needs to be known Type of signaling? DTMF? pulse? Interconnection cable wire size and capacitance will affect high frequency response, loop current, inductive pickup and pulse shaping to name just a few. The ATA requirements need to be known. A total loop resistance of 500 ohms should work, but go out to 1200 and most will fail Do you really have control over this or will you be renting a pair from the local telco? Protection should be applied on both ends for safety of the user(s) and devices. There MUST be a better way??? asterisk-us...@rogg.is wrote: Hello. I am looking for details of the maximum allowed/usable/effective wire/cable length of the connection from a FXS port of Digium analog cards to the analog telephone handset. To clarify my intention, I need to have an analog telephone connection to my asterisk box that is 3000 meters (3km) away at least. If you have any details of ATA boxes or other similar devices that I could use to do this, I'd appreciate your input. It must be able to use a regular analog telephone handset on the far end. I've searched high and low and either I'm not clever enough in using the right terms for this or it is rarely documented? Any details much appreciated. Thank you! Baldvin _ ___ -- 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 -- Dog is my co-pilot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR after SIP blind transfer.
Hi,
I can't get Asterisk to save CDRs for calls transferred via SIP blind transfer.
My extensions.conf:
[globals]
__TRANSFER_CONTEXT = transfer
[common]
exten = 123,1,Playback(demo-congrats)
exten = 123,n,Hangup()
exten = _0X.,1,Dial(SIP/${ext...@pstn-gw,60)
exten = _0X.,n,Hangup()
exten = i,1,Hangup()
exten = h,1,Hangup()
exten = t,1,Hangup()
[transfer]
exten = 123,1,Goto(common,${EXTEN},1)
Scenario A:
SIP Phone dials 123 and hangs up after 10 seconds.
CDR is recorded just fine.
Scenario B:
SIP Phone dials 02088441234 which is routed to the external peer.
After 10 seconds call is transferred (blindly) to extension 123. After
another 10 seconds external peer hangs up.
Problem: there is only one CDR recorded for the first 10 seconds long
call. Second part of the call, after 02088441234 was transferred to
123 is NOT recorded.
Is there any way to force Asterisk to record CDR in scenario B
(without using LOCAL channel)?
Regards,
Chris
___
-- 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] FXS
How about a low cost ATA? Just plug the ATA into the network, configure it - along with a SIP definition within sip.conf and you're ready to go. Lee From: Diogo Saad [mailto:diogos...@gmail.com] Sent: Tuesday, May 26, 2009 10:41 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] FXS What is the easiest way to connect my black phone to a PC running asterisk? I don't need multiple extensions, I've got just 1 phone. Is there any USB FXS adapter? Thanks -- Diogo Saad ___ -- 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] FXS
Using an ATA, Do I still need a softphone or it´s embedded in the hardware? On Tue, May 26, 2009 at 12:09 PM, Steve Edwards asterisk@sedwards.comwrote: On Tue, 26 May 2009, Diogo Saad wrote: What is the easiest way to connect my black phone to a PC running asterisk? I don't need multiple extensions, I've got just 1 phone. Is there any USB FXS adapter? An Ethernet based ATA would be more versatile. I like Digium's discontinued IAXy. Dead simple to configure, easy to travel with, no NAT headaches. Used on ebay should set you back about US$30. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Diogo Saad ___ -- 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] FXS
Diogo Saad wrote: Using an ATA, Do I still need a softphone or it´s embedded in the hardware? plain old walmart phone plugs in the ata (with or without callerid, adsi, cordless, etc) On Tue, May 26, 2009 at 12:09 PM, Steve Edwards asterisk.org http://asterisk.org@sedwards.com http://sedwards.com wrote: On Tue, 26 May 2009, Diogo Saad wrote: What is the easiest way to connect my black phone to a PC running asterisk? I don't need multiple extensions, I've got just 1 phone. Is there any USB FXS adapter? An Ethernet based ATA would be more versatile. I like Digium's discontinued IAXy. Dead simple to configure, easy to travel with, no NAT headaches. Used on ebay should set you back about US$30. Thanks in advance, Steve Edwards sedwa...@sedwards.com mailto:sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Diogo Saad ___ -- 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] Maximum cable length for analog phone from FXS port
You are correct. Telcos normally supply dial tone to business and residence for miles if there is no DSL Loading coils are used to offset the capacitance of cables, and precise spacing of these is required, and are engineered for different types of cable. John Novack David Gibbons wrote: I could be wrong but I don't think the cat5 limit of 100 meters applies to any analog signaling over that copper. I believe it only applies to Ethernet signaling. -Dave *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Danny Nicholas *Sent:* Tuesday, May 26, 2009 10:41 AM *To:* bald...@rogg.is; 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Maximum cable length for analog phone from FXS port The best a native cat5 can run is 100 meters. Unless you like paying your telco huge bucks, you should go for some kind of SIP connection to your box. *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *asterisk-us...@rogg.is *Sent:* Tuesday, May 26, 2009 9:09 AM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* [asterisk-users] Maximum cable length for analog phone from FXS port Hello. I am looking for details of the maximum allowed/usable/effective wire/cable length of the connection from a FXS port of Digium analog cards to the analog telephone handset. To clarify my intention, I need to have an analog telephone connection to my asterisk box that is 3000 meters (3km) away at least. If you have any details of ATA boxes or other similar devices that I could use to do this, I'd appreciate your input. It must be able to use a regular analog telephone handset on the far end. I've searched high and low and either I'm not clever enough in using the right terms for this or it is rarely documented? Any details much appreciated. Thank you! Baldvin ___ -- 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 -- Dog is my co-pilot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Converting Cisco 7961 to SIP
Please do! Cory J. Andrews Director New Market Initiatives Sayers Media Group VoIP Supply, LLC 454 Sonwil Drive Buffalo, NY 14225 716-250-3402 OFFICE 716-630-1548 FAX 716-601-4474 MOBILE candr...@sayersmedia.com Have I exceeded your expectations? Please share your experience with my boss, Benjamin P. Sayers, CEO NOTICE: The information contained in this email and any document attached hereto is intended only for the named recipient(s). It is the property of the VoIP Supply, LLC and shall not be used, disclosed or reproduced without the express written consent of VoIP Supply, LLC. If you are not the intended recipient, nor the employee or agent responsible for delivering this message in confidence to the intended recipient(s), you are hereby notified that you have received this transmittal in error, and any review, dissemination, distribution or copying of this transmittal or its attachments is strictly prohibited. If you have received this transmittal and/or attachments in error, please notify me immediately by reply e-mail or telephone and then delete this message, including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY 14225 USA. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: Tuesday, May 26, 2009 10:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Converting Cisco 7961 to SIP Ahh I see. In response to your other question about the auto-provisioning of Cisco phones, I wrote some scripts that work against an active directory and setup the phones automagically. I'll send the link your way if you'd like. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cory Andrews Sent: Tuesday, May 26, 2009 10:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Converting Cisco 7961 to SIP Did not mean to infer they don't perform wonderfully with Asterisk. By hack I meant that Cisco does not offer any official support for them on Asterisk. Cory J. Andrews Director New Market Initiatives Sayers Media Group VoIP Supply, LLC 454 Sonwil Drive Buffalo, NY 14225 716-250-3402 OFFICE 716-630-1548 FAX 716-601-4474 MOBILE candr...@sayersmedia.com Have I exceeded your expectations? Please share your experience with my boss, Benjamin P. Sayers, CEO NOTICE: The information contained in this email and any document attached hereto is intended only for the named recipient(s). It is the property of the VoIP Supply, LLC and shall not be used, disclosed or reproduced without the express written consent of VoIP Supply, LLC. If you are not the intended recipient, nor the employee or agent responsible for delivering this message in confidence to the intended recipient(s), you are hereby notified that you have received this transmittal in error, and any review, dissemination, distribution or copying of this transmittal or its attachments is strictly prohibited. If you have received this transmittal and/or attachments in error, please notify me immediately by reply e-mail or telephone and then delete this message, including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY 14225 USA. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: Tuesday, May 26, 2009 10:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Converting Cisco 7961 to SIP Cory, Precisely what do you mean by 'Anything other than Callmanager will essentially be a hack'? I've got nearly 100 Cisco 79x1s in production using Asterisk against the SIP image. They're not 'hacked', they're set up properly against the Cisco provided SIP image and are rock-solid stable. I would pit them against any of the cheaper model SIP phones any time, any place, any day. I've written scripts to do nearly everything that call manager can do without paying hundreds of dollars per user for the call manager software. Just about the only thing they can't do at the moment is BLF because they require SIP over TCP to handle SIP messages about BLF status, something that I'm not willing to implement just yet. In the past, Cisco phones have had a bad rap as not being usable outside of a call manager environment. That's just not the case. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cory Andrews Sent: Tuesday, May 26, 2009 9:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Converting Cisco 7961 to SIP Darrin, The files you are using are consistent with SIP for Cisco Call Manager. Anything other than Callmanager will
Re: [asterisk-users] Maximum cable length for analog phone from FXS port
Miguel Molina wrote: randulo escribió: On Tue, May 26, 2009 at 5:31 PM, Danny Nicholas da...@debsinc.com wrote: I run my analog telco over cat5, but that's in-house and definitely not 3km. That sounds really far for current loop stuff. I was doing that too. I asked this same question a few years ago and the answer was 100-200 meters. This is just a quick rule of thumb, but it seems about right. 3km, I doubt that would work, but it depends, as someone said, totally depending on ohm's law :) What I think about this is, the length of the copper cable between the central office and home is usually several km, but definitely helped by the central office circuitry (current source instead of voltage source, that guarantees a minimum ringing voltage on the far end). What I don't know is, a FXS port behaves the same as a central office, electrically speaking? If that is so, you could extend your 3km of cable without problems, but I think you can have some noise problems depending on what places the cable has to go through. from the ringing point of view, the CO ring generator is usually truly a sine wave and this propagates well through a cable. The cheap fxs ports are mostly square waves and lower voltages with limited current sourcing (check the REN numbers they are capable of ringing for a comparison if its listed) some of the cheap ones have trouble ringing a phone plugged in with a short line cord. So in addition to being frequency choked by the long run the square wave will get reduced in amplitude, and it may well have been marginal amplitude to begin with. so depending what fxs hardware you have driving it and the load from the phone, results will range from works perfectly to does not work at all. r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57 1)6500816 Móvil: (+57)3138873587 ___ -- 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] A problem in playing sound files
On Tue, 26 May 2009, abdelkader wrote: I have 8 DID: 7 from a provider1 and 1 from provider2. Each time a customer calls one of the DID, the system plays a message. The problem is that the message is played normally for all the DIDs from the provider1 and is not played (not heard) for the DID from provider2. Codecs would be a good place to start. Maybe NAT issues. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maximum cable length for analog phone from FXS port
On Tue, 26 May 2009, Danny Nicholas wrote: I run my analog telco over cat5, but that's in-house and definitely not 3Km. Of course - and that is just fine. If you were running ethernet signalling over that CAT5 than your 100m limit would apply. If you were running gigabit over that same cable its more like 80 feet. He isn't asking about ethernet signalling. As many posts have shown this morning, the actual length limit for running analog voice and DTMF signalling over the cable depends on many things that the OP probably has no control over. YMMV, like most things. j -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Tuesday, May 26, 2009 10:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: bald...@rogg.is Subject: Re: [asterisk-users] Maximum cable length for analog phone from FXS port On Tue, 26 May 2009, Danny Nicholas wrote: The best a native cat5 can run is 100 meters. Unless you like paying your telco huge bucks, you should go for some kind of SIP connection to your box. He was asking about an analog telco connection - not an ethernet drop. j _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk-us...@rogg.is Sent: Tuesday, May 26, 2009 9:09 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Maximum cable length for analog phone from FXS port Hello. I am looking for details of the maximum allowed/usable/effective wire/cable length of the connection from a FXS port of Digium analog cards to the analog telephone handset. To clarify my intention, I need to have an analog telephone connection to my asterisk box that is 3000 meters (3km) away at least. If you have any details of ATA boxes or other similar devices that I could use to do this, I'd appreciate your input. It must be able to use a regular analog telephone handset on the far end. I've searched high and low and either I'm not clever enough in using the right terms for this or it is rarely documented? Any details much appreciated. Thank you! Baldvin ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP over VPN
Assuming you mean the firewall in front of the client, you don't need to open any ports as long as the VPN client is tunneling all traffic to and from the Asterisk server. I set NAT=yes in the config file for the extensions behind a VPN. -Dave From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marco Sambo Sent: Tuesday, May 26, 2009 11:21 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] SIP over VPN Hi all, I have a question. I have a VPN and I want to use a SIP softphone on my notebook using with asterisk. But I have some problem with firewall and port. Someone knows which ports I should open on my firewall??? I can't connect ... Thanks all. Marco ___ -- 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] Maximum cable length for analog phone from FXS port
Excellent analysis of the real world. Start with this, and work out the issues, or go to VOIP. Cary Fitch _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Wilton Helm Sent: Tuesday, May 26, 2009 11:33 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Maximum cable length for analog phone fromFXS port There are a lot of factors that impact this. First, CAT 5, while usable is overkill. Cat 3 (otherwise known as I/O wire) works equally well for voice grade lines. That being said, for that long a run, a heavier gauge wire would be better. I believe telcos use 18 - 22 guage (Cat 5 and Cat 3 are both 26 awg). This has less resistive loss. Most FXS or ATA devices use 24 volts or less for battery. That works fine for short loops, but limits the range. A central office POTS port normally uses 48 VDC which works well to several KM. If the customer is at the end of a long run in a rural area, they use a long line card which uses 75 volts. (In rural communities, they often place the line cards in a roadside remote terminal and use statistically multiplexed T1s to make it appear to the switch as a part of it. That addresses the DC characteristics, which can be reduced to ohms law. A phone needs around 8 V @ .02 A. The wire resistance determine the drop (E = IR) and the source voltage determines whether there will be enough left. The A.C. characteristics are more complicated. The FXS must do a 2 wire to 4 wire conversion, which involves matching the impedance of the line. The FXS is generally designed for relatively short lines, so might not be able to match either the resistance or capacitance found in a long run. Heavier wire will minimize this. In addition to that, the transmit side of the 2 wire to 4 wire circuit must be able to drive the load it sees, and again it may not be designed with a long run in mind. Finally, COs line cards have the ability to adjust receive and transmit gain to compensate for sound level losses in long lines. While this isn't routinely done on simple circuits, it is an option an FXS doesn't generally have. In addition, the more gain that is inserted, the harder it is to balance to 2 wire to 4 wire circuit, and the more complex it has to be in order to support this. Wilton ___ -- 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] FXS
how do I configure my SIP account information? I mean, sip proxy and etc. On Tue, May 26, 2009 at 1:19 PM, Jon Pounder j...@inline.net wrote: Diogo Saad wrote: Using an ATA, Do I still need a softphone or it´s embedded in the hardware? plain old walmart phone plugs in the ata (with or without callerid, adsi, cordless, etc) On Tue, May 26, 2009 at 12:09 PM, Steve Edwards asterisk.org http://asterisk.org@sedwards.com http://sedwards.com wrote: On Tue, 26 May 2009, Diogo Saad wrote: What is the easiest way to connect my black phone to a PC running asterisk? I don't need multiple extensions, I've got just 1 phone. Is there any USB FXS adapter? An Ethernet based ATA would be more versatile. I like Digium's discontinued IAXy. Dead simple to configure, easy to travel with, no NAT headaches. Used on ebay should set you back about US$30. Thanks in advance, Steve Edwards sedwa...@sedwards.com mailto:sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Diogo Saad ___ -- 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 -- Diogo Saad ___ -- 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] Maximum cable length for analog phone from FXS port
There are a lot of factors that impact this. First, CAT 5, while usable is overkill. Cat 3 (otherwise known as I/O wire) works equally well for voice grade lines. That being said, for that long a run, a heavier gauge wire would be better. I believe telcos use 18 - 22 guage (Cat 5 and Cat 3 are both 26 awg). This has less resistive loss. Most FXS or ATA devices use 24 volts or less for battery. That works fine for short loops, but limits the range. A central office POTS port normally uses 48 VDC which works well to several KM. If the customer is at the end of a long run in a rural area, they use a long line card which uses 75 volts. (In rural communities, they often place the line cards in a roadside remote terminal and use statistically multiplexed T1s to make it appear to the switch as a part of it. That addresses the DC characteristics, which can be reduced to ohms law. A phone needs around 8 V @ .02 A. The wire resistance determine the drop (E = IR) and the source voltage determines whether there will be enough left. The A.C. characteristics are more complicated. The FXS must do a 2 wire to 4 wire conversion, which involves matching the impedance of the line. The FXS is generally designed for relatively short lines, so might not be able to match either the resistance or capacitance found in a long run. Heavier wire will minimize this. In addition to that, the transmit side of the 2 wire to 4 wire circuit must be able to drive the load it sees, and again it may not be designed with a long run in mind. Finally, COs line cards have the ability to adjust receive and transmit gain to compensate for sound level losses in long lines. While this isn't routinely done on simple circuits, it is an option an FXS doesn't generally have. In addition, the more gain that is inserted, the harder it is to balance to 2 wire to 4 wire circuit, and the more complex it has to be in order to support this. Wilton ___ -- 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] Suggest good calling service for London
Hello All, We are setting up call center of 10 agents and expecting its growth till 30 agents. Mainly calling is within UK. Please suggest some good service for UK dialing with London DID. Regards, Kashif Naeem Business Development Manager Hadi Telecom www.haditelecom.com Cell: +92 (0)345 4226006 Office: +92 (0)42 5692766 Email: kas...@haditelecom.com MSN: kashif__na...@hotmail.com Gmail: meet.kas...@gmail.com Skype: kashif.naeem 302 Y Commercial Area, 2nd Floor DHA Lahore, Pakistan. ___ -- 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] Maximum cable length for analog phone from FXS port
Wilton Helm wrote: one thing I missed mentioning about fxs devices - the linksys/sipura ones actually allow you to set line characteristics on the slic inside it. you can vary from the 600ohm default, and tweak gains a bit. Some mix of a capacitive line or different resistance may help. never tried myself but there are a ton of things you can play with. There are a lot of factors that impact this. First, CAT 5, while usable is overkill. Cat 3 (otherwise known as I/O wire) works equally well for voice grade lines. That being said, for that long a run, a heavier gauge wire would be better. I believe telcos use 18 – 22 guage (Cat 5 and Cat 3 are both 26 awg). This has less resistive loss. Most FXS or ATA devices use 24 volts or less for “battery”. That works fine for short loops, but limits the range. A central office POTS port normally uses 48 VDC which works well to several KM. If the customer is at the end of a long run in a rural area, they use a “long line” card which uses 75 volts. (In rural communities, they often place the line cards in a roadside “remote terminal” and use statistically multiplexed T1s to make it appear to the switch as a part of it. That addresses the DC characteristics, which can be reduced to ohms law. A phone needs around 8 V @ .02 A. The wire resistance determine the drop (E = IR) and the source voltage determines whether there will be enough left. The A.C. characteristics are more complicated. The FXS must do a 2 wire to 4 wire conversion, which involves matching the impedance of the line. The FXS is generally designed for relatively short lines, so might not be able to match either the resistance or capacitance found in a long run. Heavier wire will minimize this. In addition to that, the transmit side of the 2 wire to 4 wire circuit must be able to drive the load it sees, and again it may not be designed with a long run in mind. Finally, COs line cards have the ability to adjust receive and transmit gain to compensate for sound level losses in long lines. While this isn’t routinely done on simple circuits, it is an option an FXS doesn’t generally have. In addition, the more gain that is inserted, the harder it is to balance to 2 wire to 4 wire circuit, and the more complex it has to be in order to support this. Wilton ___ -- 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] Maximum cable length for analog phone from FXS port
You are exactly right. Cat 5 had no advantage over cheaper wire for voice, and the length limitations are meaningless. Consider that Cat 5 is typically use with signals that extent to 30 MHz or beyond. A voice grade analog circuit must go to 4 KHz (1/10,000 as much). At 4 KHz, the wire generally doesn't even act like a controlled impedance. Wilton From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: Tuesday, May 26, 2009 8:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Maximum cable length for analog phone from FXS port I could be wrong but I don't think the cat5 limit of 100 meters applies to any analog signaling over that copper. I believe it only applies to Ethernet signaling. -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
Re: [asterisk-users] FXS
Diogo Saad wrote: how do I configure my SIP account information? I mean, sip proxy and etc. you need just a couple pieces of information server (put this in any setting that says proxy or host etc, all set the same) account (the extension in asterisk, put anywhere that sounds like a non-display only field) password (secret, key, password etc., should be one field that takes this in the config) register = yes basically that's it. you mean need to disable feature codes etc, but the above will get most any sip device working with asterisk once you setup an extension for it. On Tue, May 26, 2009 at 1:19 PM, Jon Pounder j...@inline.net mailto:j...@inline.net wrote: Diogo Saad wrote: Using an ATA, Do I still need a softphone or it´s embedded in the hardware? plain old walmart phone plugs in the ata (with or without callerid, adsi, cordless, etc) On Tue, May 26, 2009 at 12:09 PM, Steve Edwards asterisk.org http://asterisk.org http://asterisk.org@sedwards.com http://sedwards.com http://sedwards.com wrote: On Tue, 26 May 2009, Diogo Saad wrote: What is the easiest way to connect my black phone to a PC running asterisk? I don't need multiple extensions, I've got just 1 phone. Is there any USB FXS adapter? An Ethernet based ATA would be more versatile. I like Digium's discontinued IAXy. Dead simple to configure, easy to travel with, no NAT headaches. Used on ebay should set you back about US$30. Thanks in advance, Steve Edwards sedwa...@sedwards.com mailto:sedwa...@sedwards.com mailto:sedwa...@sedwards.com mailto:sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Diogo Saad ___ -- 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 -- Diogo Saad ___ -- 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] Maximum cable length for analog phone from FXS port
Wilton Helm wrote: You are exactly right. Cat 5 had no advantage over cheaper wire for voice, and the length limitations are meaningless. Consider that Cat 5 is typically use with signals that extent to 30 MHz or beyond. A voice grade analog circuit must go to 4 KHz (1/10,000 as much). At 4 KHz, the wire generally doesn’t even act like a controlled impedance. I completely agree, but that said I still use cat5/e for everything anyway, not worth having more than one kind of wire, and lets you change your mind later on the usage. Once you are using outside plant facilities though, you live with what you get, and don't expect much. Wilton *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Gibbons *Sent:* Tuesday, May 26, 2009 8:50 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Maximum cable length for analog phone from FXS port I could be wrong but I don’t think the cat5 limit of 100 meters applies to any analog signaling over that copper. I believe it only applies to Ethernet signaling. -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 ___ -- 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] Maximum cable length for analog phone from FXS port
I run my analog telco over cat5, but that's in-house and definitely not 3Km. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Tuesday, May 26, 2009 10:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: bald...@rogg.is Subject: Re: [asterisk-users] Maximum cable length for analog phone from FXS port On Tue, 26 May 2009, Danny Nicholas wrote: The best a native cat5 can run is 100 meters. Unless you like paying your telco huge bucks, you should go for some kind of SIP connection to your box. He was asking about an analog telco connection - not an ethernet drop. j _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk-us...@rogg.is Sent: Tuesday, May 26, 2009 9:09 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Maximum cable length for analog phone from FXS port Hello. I am looking for details of the maximum allowed/usable/effective wire/cable length of the connection from a FXS port of Digium analog cards to the analog telephone handset. To clarify my intention, I need to have an analog telephone connection to my asterisk box that is 3000 meters (3km) away at least. If you have any details of ATA boxes or other similar devices that I could use to do this, I'd appreciate your input. It must be able to use a regular analog telephone handset on the far end. I've searched high and low and either I'm not clever enough in using the right terms for this or it is rarely documented? Any details much appreciated. Thank you! Baldvin ___ -- 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] Maximum cable length for analog phone from FXS port
On Tue, May 26, 2009 at 12:44:26PM -0400, Jon Pounder wrote: Wilton Helm wrote: one thing I missed mentioning about fxs devices - the linksys/sipura ones actually allow you to set line characteristics on the slic inside it. you can vary from the 600ohm default, and tweak gains a bit. Some mix of a capacitive line or different resistance may help. never tried myself but there are a ton of things you can play with. Any of those are actually important? Which? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maximum cable length for analog phone from FXS port
Tzafrir Cohen wrote: On Tue, May 26, 2009 at 05:39:46PM +0200, randulo wrote: On Tue, May 26, 2009 at 5:31 PM, Danny Nicholas da...@debsinc.com wrote: I run my analog telco over cat5, but that's in-house and definitely not 3km. That sounds really far for current loop stuff. I was doing that too. I asked this same question a few years ago and the answer was 100-200 meters. This is just a quick rule of thumb, but it seems about right. 3km, I doubt that would work, but it depends, as someone said, totally depending on ohm's law :) If it were to depend solely on Ohm's law, than 3km would be marginal but probably within reach. Not exactly sure what other factors are there to count. The copper planning limit (or whatever local term your telco uses) is typically 10km. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bandwidth management and ADSL router
Thanks for all. But what all gave me was a software need to be installed on PC, but I am looking for a router (ADSL router) that can does this, because usually the ADSL router is the default gateway where all the traffic goes out and in. Any ADSL router device can do this? About Draytek, as I understand that control can be done only at upload traffic and not download traffic, while 90% of the problem are coming from download traffic, so this is not the needed. Any advise in that direction? Regards Bilal ___ -- 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] Maximum cable length for analog phone from FXS port
For long distances, a wireless point-to-point might be more economical than trenching. e.g: Carlson Wideband CDMA Spread Spectrum Phone Line Extender http://www.oksolar.com/communications/phone_line_ext.htm ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP over VPN
Hi all, I have a question. I have a VPN and I want to use a SIP softphone on my notebook using with asterisk. But I have some problem with firewall and port. Someone knows which ports I should open on my firewall??? I can't connect ... Thanks all. Marco ___ -- 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] Maximum cable length for analog phone from FXS port
On Tue, 26 May 2009, Danny Nicholas wrote: The best a native cat5 can run is 100 meters. Unless you like paying your telco huge bucks, you should go for some kind of SIP connection to your box. He was asking about an analog telco connection - not an ethernet drop. j _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk-us...@rogg.is Sent: Tuesday, May 26, 2009 9:09 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Maximum cable length for analog phone from FXS port Hello. I am looking for details of the maximum allowed/usable/effective wire/cable length of the connection from a FXS port of Digium analog cards to the analog telephone handset. To clarify my intention, I need to have an analog telephone connection to my asterisk box that is 3000 meters (3km) away at least. If you have any details of ATA boxes or other similar devices that I could use to do this, I'd appreciate your input. It must be able to use a regular analog telephone handset on the far end. I've searched high and low and either I'm not clever enough in using the right terms for this or it is rarely documented? Any details much appreciated. Thank you! Baldvin ___ -- 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] Bandwidth management and ADSL router
A lot of ISP adsl modems aren't capable. But most should be. Just log in to it give it a private IP for your lan(192.X.X.X or whatever your using) then have all your computers use that local IP as their gateway address. If you have an ADSL modem which doesn't then simple get a router (hell a Linksys/Dlink $50 cheapy from wallmart would work) and have the ADSL plug into the router and all the stations use the router for their gateway. If you have a spare server or virtual server space you can use Vyatta (Vyatta.com) it is a free open source router/firewall/vpn/few other things. I've never used it in a virtual environment, but I see no reason why it wouldn't work that way. Also note that it requires almost nothing to run so you can put it on an old 1Ghz machine and It would still operate just fine. James Shigley Monroe Telephone Answering Service 409-981-9213 Infinity 5.5,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. Common sense is the collection of prejudices acquired by age eighteen. -- Albert Einstein Once you can accept the universe as matter expanding into nothing that is something,wearing stripes with plaid comes easy. -- Albert Einstein Theory is when you know something, but it doesn't work. Practice is when something works, but you don't know why. Programmers combine theory and practice: Nothing works and they don't know why.-Anonymous Developer -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Tuesday, May 26, 2009 11:55 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Bandwidth management and ADSL router Thanks for all. But what all gave me was a software need to be installed on PC, but I am looking for a router (ADSL router) that can does this, because usually the ADSL router is the default gateway where all the traffic goes out and in. Any ADSL router device can do this? About Draytek, as I understand that control can be done only at upload traffic and not download traffic, while 90% of the problem are coming from download traffic, so this is not the needed. Any advise in that direction? Regards Bilal ___ -- 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] No Voice - only noisy audio
Hi Folks, I'm trying to use my mobile as a trunk via bluetooth - calls done in a softphone go thru GSM network and calls destinated to my mobile are answered at the softphone. I have asterisk configured to do so but I'm facing an issue - Audio is audible but it’s not intelligible. I feel like the audio is breaking. Below is the asterisk log. I also get lots of “hci_scodata_packet: hci0 SCO packet for unknown connection handle X” and btusb_isoc_complete: hci0 corrupted SCO packet entries in kernel logs. Can anybody please help? Tks ++ 13:37:17 chan_sip.c: Allocating new SIP dialog for 42eb60ff0430607e7eb97cc86...@192.168.0.204 - OPTIONS (No RTP) 13:37:17 acl.c: Found IP address for this socket 13:37:17 chan_sip.c: Initializing initreq for method OPTIONS - callid 5bae8a561541036e45990a137366c...@192.168.0.204 13:37:17 chan_sip.c: Trying to put 'OPTIONS si' onto UDP socket destined for 192.168.0.84:27928 13:37:17 chan_sip.c: Stopping retransmission on ' 5bae8a561541036e45990a137366c...@192.168.0.204' of Request 102: Match Found 13:37:17 chan_sip.c: Destroying SIP dialog 5bae8a561541036e45990a137366c...@192.168.0.204 13:37:40 acl.c: Found IP address for this socket 13:37:40 netsock.c: == Using SIP RTP CoS mark 5 13:37:40 chan_sip.c: Setting NAT on RTP to Off 13:37:40 chan_sip.c: Allocating new SIP dialog for N2ExYWUwNGJkODAzZGMyYjBmYzQwMTY1YzgwMGQ5MWM. - INVITE (With RTP) 13:37:40 chan_sip.c: Received INVITE (5) - Command in SIP INVITE 13:37:40 chan_sip.c: Setting NAT on RTP to Off 13:37:40 chan_sip.c: Trying to put 'SIP/2.0 40' onto UDP socket destined for 192.168.0.84:27928 13:37:40 chan_sip.c: Received ACK (6) - Command in SIP ACK 13:37:40 chan_sip.c: Stopping retransmission on 'N2ExYWUwNGJkODAzZGMyYjBmYzQwMTY1YzgwMGQ5MWM.' of Response 1: Match Found 13:37:40 chan_sip.c: Received INVITE (5) - Command in SIP INVITE 13:37:40 chan_sip.c: Setting NAT on RTP to Off 13:37:40 chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) 13:37:40 chan_sip.c: Checking SIP call limits for device 1000 13:37:40 chan_sip.c: Updating call counter for incoming call 13:37:40 devicestate.c: No provider found, checking channel drivers for SIP - 1000 13:37:40 chan_sip.c: Checking device state for peer 1000 13:37:40 devicestate.c: Changing state for SIP/1000 - state 2 (In use) 13:37:40 devicestate.c: device 'SIP/1000' state '2' 13:37:40 app_queue.c: Device 'SIP/1000' changed to state '2' (In use) but we don't care because they're not a member of any queue. 13:37:40 chan_sip.c: *** Our native formats are 0x4 (ulaw) 13:37:40 chan_sip.c: *** Joint capabilities are 0xc (ulaw|alaw) 13:37:40 chan_sip.c: *** Our capabilities are 0xe (gsm|ulaw|alaw) 13:37:40 chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) 13:37:40 chan_sip.c: This channel will not be able to handle video. 13:37:40 chan_sip.c: build_route: Contact hop: sip:1...@192.168.0.84:27928 13:37:40 chan_sip.c: SIP/1000-0021a568: New call is still down Trying... 13:37:40 chan_sip.c: Trying to put 'SIP/2.0 10' onto UDP socket destined for 192.168.0.84:27928 13:37:40 devicestate.c: No provider found, checking channel drivers for SIP - 1000 13:37:40 chan_sip.c: Checking device state for peer 1000 13:37:40 devicestate.c: Changing state for SIP/1000 - state 2 (In use) 13:37:40 devicestate.c: device 'SIP/1000' state '2' 13:37:40 app_queue.c: Device 'SIP/1000' changed to state '2' (In use) but we don't care because they're not a member of any queue. 13:37:40 pbx.c: Launching 'Answer' 13:37:40 ] pbx.c: -- Executing [1...@from-internal:1] Answer(SIP/1000-0021a568, ) in new stack 13:37:40 devicestate.c: No provider found, checking channel drivers for SIP - 1000 13:37:40 chan_sip.c: Checking device state for peer 1000 13:37:40 devicestate.c: Changing state for SIP/1000 - state 2 (In use) 13:37:40 devicestate.c: device 'SIP/1000' state '2' 13:37:40 app_queue.c: Device 'SIP/1000' changed to state '2' (In use) but we don't care because they're not a member of any queue. 13:37:40 chan_sip.c: SIP answering channel: SIP/1000-0021a568 13:37:40 chan_sip.c: Setting framing from config on incoming call 13:37:40 chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True 13:37:40 chan_sip.c: ** Our prefcodec: 0x0 (nothing) 13:37:40 chan_sip.c: -- Done with adding codecs to SDP 13:37:40 channel.c: Internal timing is disabled (option_internal_timing=0 chan-timingfd=28) 13:37:40 chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) 13:37:40 chan_sip.c: Trying to put 'SIP/2.0 20' onto UDP socket destined for 192.168.0.84:27928 13:37:40 rtp.c: Got RTCP report of 132 bytes 13:37:40 pbx.c: Launching 'Dial' 13:37:40 ] pbx.c: -- Executing [1...@from-internal:2] Dial(SIP/1000-0021a568, Mobile/Carlos/909037079681) in new stack 13:37:40 rtp.c: Channel 'Mobile/Carlos-0213' has no RTP, not doing anything 13:37:40 channel.c: Not copying variable DIALEDTIME. 13:37:40 channel.c: Not copying variable ANSWEREDTIME.
[asterisk-users] STUN setting in Asterisk 1.6.X
I have been trying out several stun servers with Asterisk 1.6.0.9 and 1.6.1.0 and I keep getting the following message: [May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor: stun failed [May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor: stun failed [May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor: stun failed [May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor: stun failed [May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor: stun failed [May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor: stun failed No matter which STUN server I point to I get those messages. Am I missing some other setting? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ -- 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 and Data Modem
Jon Morgan wrote:
Hi All,
We have an Asterisk 1.4.21.2 box which uses a 2 port Digium card to bridge
calls, as follows:
ISDN Provider --- Span 1(pri_cpe) --- Span 2(pri_net) Phone
System
The company that looks after our internal phone system can no longer dial in
using their data modem in order to maintain the internal phone system. Is
there any way we can configure our asterisk to allow them to dial in using
their modem?
Regards,
Jon.
___
-- 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
Hi jon
What system is it?
you need to set the transfer capability
eg
exten = _X.,1,gotoif($[${TRANSFERCAPABILITY}=DIGITAL]?8:)
exten = _X.,2,Noop
exten = _X.,3,ringing
exten = _X.,4,set(CDR(accountcode)=${EXTEN})
exten = _X.,5,Noop
exten = _X.,6,dial(ZAP/g2/${EXTEN},,r)
exten = _X.,7,hangup
exten = _X.,8,Set(CHANNEL(transfercapability)=DIGITAL)
exten = _X.,9,dial(ZAP/g2/${EXTEN})
exten = _X.,n,hangup
Regards
Robb
___
-- 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] Bandwidth management and ADSL router
I've had good luck using a sangoma S518 ADSL card in a linux box. the logging capabilities are supurb (cought my provider not providing what they said they were and great for troubleshooting as it logs line speed and dropouts to the second). support is also top notch. once installed it looks to the system like any other interface. Since it looks to the system like any other interface you have the full power of routing, bridging, firewalling, iptables, neumerous queing schemes, etc. everything linux has to offer. It has served me well and is extremely flexable. Eric Fort FortConsulting On Tue, May 26, 2009 at 4:32 AM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; I discover that most of the voice cutting complain are coming from the Internet bandwidth when we are connecting two remote offices togethor via Asterisk or any other IP PBX. Anyone has an idea on a ADSL router that work as ADSL + Bandwidth division? So we can resolve the problem of providing a guaranteed bandwidth for the voice packets instead of suffering the voice cutting? Regards Bilal ___ -- 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] Strange message in CLI
While I was in the console looking for something else, this appeared when I called in on my cell. [May 26 12:17:26] NOTICE[3364]: chan_sip.c:17229 handle_request_invite: Sending fake auth rejection for user xxx xxx xx sip:xxx...@xxx.xxx.xx.xxx;tag=as04e93fb9 What does this mean? Searching the net simply brought me to the source files. Thanks! jlc ___ -- 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] Silly (??) question about chan_dahdi
Hi, these are my first steps with DAHDI and I finally managed to get asterisk to load chan_dahdi (after I found out, that I need libpri). But how do I tell chan_dahdi on which isdn numbers it should react? I haven't found a parameter like incomingmsn for chan_capi in the documentation. Thanks for your help, Stefan ___ -- 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] Add Monitor application to call suppresses audio
On Wed, May 13, 2009 at 11:53 AM, Barry L. Kline blkl...@attglobal.net wrote:
If I insert a Monitor() prior to dialing the outbound call, I get no
audio in the recording and the caller hears no audio. Occasionally it
works (perhaps 1 out of 5 times) but most of the time the caller can't
hear the callee, and vice versa.
The fully working code looks like this:
1) exten = s,n(place),Verbose(4,Dialing answering service);
2) exten = s,n,Playback(vrec_prompts/this-call-may-be-recorded);
3) exten = s,n,Set(GROUP()=ANSSVC);
4) exten =
s,n,Set(CALLFILENAME=${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}_${CALLERID(num)});
5) exten = s,n,Dial(${OUTGOING_PRI}/${ANSWERINGSVC},15,r);
6) exten = s,n,Goto(s-${DIALSTATUS},1);
What is the 6 for?
What is the goto supposed to do?
This could certainly explain why the first call works and not the
subsequent calls.
Why don't you want to just hangup the call after 5 completes?
___
-- 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] Silly (??) question about chan_dahdi
You define context= for the channels in dahdi.conf and then in extensions.conf you define those numbers in that particular context name eg: dahdi.conf context=incoming channel = 1-15,17-31 extensions.conf [incoming] exten = _X.,1,Answer exten = _X.,2,Echo and it will react to all numbers that come on that circuit and do Echo app on incoming calls Martin On Tue, May 26, 2009 at 1:30 PM, Stefan-Michael Guenther asteris...@in-put.de wrote: Hi, these are my first steps with DAHDI and I finally managed to get asterisk to load chan_dahdi (after I found out, that I need libpri). But how do I tell chan_dahdi on which isdn numbers it should react? I haven't found a parameter like incomingmsn for chan_capi in the documentation. Thanks for your help, Stefan ___ -- 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] STUN setting in Asterisk 1.6.X
[May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor: stun failed No matter which STUN server I point to I get those messages. Am I missing some other setting? Hey Carlos, That just means the stun request failed, there are several reasons for that, I won't even try to guess. So, first try this on the Asterisk CLI: stun set debug on That should give you (and us) more information to troubleshoot why the stun request failed (also enable debug and verbosity as usual). -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.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] Bug or feature in 1.6.1 (Was: How to register with TCP transport) ?
Hi, Digging on this case : 2009/5/26 Olivier oza-4...@myamail.com Hi, In my sip.conf, I've got : [general](+) ; register=tcp://trunk4ipbx:passw...@192.168.100.129trunk4ipbx%3apassw...@192.168.100.129 register=trunk4ipbx:passw...@192.168.100.129trunk4ipbx%3apassw...@192.168.100.129 When I'm using the TCP line instead of the other, I've got : [May 26 17:58:42] NOTICE[2859]: chan_sip.c:20169 sip_parse_host: '/' is not a valid port number on line 25 of sip.conf. using default. [May 26 17:58:42] WARNING[2859]: chan_sip.c:6560 sip_register: Format for registration is [transport://]user[:secret[:authuse...@domain[:port][/extension][~expiry] at line 25 Is this register=tcp://trunk4ipbx:passw...@192.168.100.129trunk4ipbx%3apassw...@192.168.100.129 statement correct ? Regards I read in chan_sip.c that block inside sip_register : /* split [/contact][~expiry] */ expire = strchr(buf, '~'); if (expire) *expire++ = '\0'; callback = strrchr(buf, '/');// My comment: contact is search at the end of input register line if (callback) *callback++ = '\0'; if (ast_strlen_zero(callback)) callback = s; sip_parse_host(buf, lineno, username, portnum, transport); Given an input line such as register=tcp:// trunk4ipbx:passw...@192.168.100.129 trunk4ipbx%3apassw...@192.168.100.129, register line is truncated as the last occurence of '/' is the tcp:// string. When commenting out this callback = strrchr(buf, '/'); , input line register=tcp://trunk4ipbx:passw...@192.168.100.129trunk4ipbx%3apassw...@192.168.100.129 seems to be processed appropriately. My question is is this legal to input register lines without any /contact field ? If positive, then there is a bug is 1.6.1. If negative, would you agree to have a more appropriate logging than sip_parse_host: '/' is not a valid port number ... ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Indications.conf and tone generation volume
Sorry if this is a repost - I never saw a copy of this go out last week. Can anyone tell me if there is a way to vary the output levels (dB) of the tones generated in indications.conf? I generate a few custom tones and sometimes people tell me they are a little too loud. Thanks Lee ___ -- 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] Maximum cable length for analog phone from FXS port
On Tue, 2009-05-26 at 10:26 -0400, John Novack wrote: That is a pretty long run. The type of analog phone can be an issue. How LITTLE loop current will it operate on? Most need more than 20 Ma to signal properly, and the voltage output of the ATA needs to be known Type of signaling? DTMF? pulse? Interconnection cable wire size and capacitance will affect high frequency response, loop current, inductive pickup and pulse shaping to name just a few. The ATA requirements need to be known. A total loop resistance of 500 ohms should work, but go out to 1200 and most will fail Do you really have control over this or will you be renting a pair from the local telco? Protection should be applied on both ends for safety of the user(s) and devices. There MUST be a better way??? I would suggest making a wifi connection with directional hi-gain antenna's. Ans a small box at the other end. Have a look at: http://www.fit-pc.net/fitpc-2-p-2.html or http://www.fit-pc.info/downloads/handleidingen/fit_pc_2_eng.pdf ___ -- 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] Add Monitor application to call suppresses audio
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
David Backeberg wrote:
5) exten = s,n,Dial(${OUTGOING_PRI}/${ANSWERINGSVC},15,r);
6) exten = s,n,Goto(s-${DIALSTATUS},1);
What is the 6 for?
What is the goto supposed to do?
Hi David.
The '6' is in case I get a CHANUNAVAIL or other error back from the
Dial command. If the call is connected then I never get to '6'.
I have determined that the only calls I seem to be having trouble
monitoring are the ones sent to my answering service. If I terminate
the call to my cell phone, my home POTS line, a POTS line here in the
office or even to the inbound PRI at the office, things work fine. I
can even record calls to the answering service's published number. It's
just when I go to the number assigned to us that there is trouble and
I'm currently chasing down the owner of that service to see exactly what
I'm dropping into there.
Thanks!
Barry
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.5 (GNU/Linux)
iD8DBQFKHEbRCFu3bIiwtTARAvlIAJ0Se61+0k6W3ixwZOm8/Sz+ixZqXQCgqLnz
2kLwyY8bHLrs/aaGd9nrho8=
=Tbri
-END PGP SIGNATURE-
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] multiple bind ports with TCP and UDP
Hi, In this thread http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/223399/focus=223401, one conclusion was that an easy way to set 2 different trunks with different binding ports was to use TCP and UDP transport. serverA udp:5060- serverB | | tcp:5062 Has anyone successfully set this up ? I'm using 1.6.1 and I've trouble to do this. Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fax Machines across carrier SIP trunk? General recommendation?
Customer has a Verizon Business SIP trunk, I'm still used to PRI T1 myself for local service. The fax machines are having some issues (I can use analog phone to call out fine) and I'm checking on modem passthrough with Verizon, but wonder if any else is using Verizon Business for SIP trunk and what your faxing milage was? Did they support G711 and modem-passthough, etc? Also checking QoS, etc. - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you.___ -- 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] h extension and channel variables
Thomas Kenyon dig...@sanguinarius.co.uk writes: In that case if there is an intervening call that is shorter, then the $calledID will be wrong. That isn't how Asterisk variables work. They aren't global to all calls, they are local to the call you happen to be in. So no, an intervening call won't cause problems. /Benny ___ -- 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] Fax Machines across carrier SIP trunk? General recommendation?
Modem and/or analog passthrough over SIP trunk, not on a LAN? I wouldn't bother - it doesn't even work very well on an uncontended LAN due to excessive jitter, let alone over the Internet or semi-private Layer 2 cloud product. T.38 or bust. The other's fax mileage is measured in gallons per mile, not miles per gallon. Jason Aarons (US) wrote: Customer has a Verizon Business SIP trunk, I’m still used to PRI T1 myself for local service. The fax machines are having some issues (I can use analog phone to call out fine) and I’m checking on modem passthrough with Verizon, but wonder if any else is using Verizon Business for SIP trunk and what your faxing milage was? Did they support G711 and modem-passthough, etc? Also checking QoS, etc. * Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. * ___ -- 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- 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
