[asterisk-users] Polycom SoundPoint IP 650 freezes on boot after adding just one custom ringtone
Hi I'm new to this list, so please forgive me off-topic or RTFM-questions. I have an asterisk/elastix driven phone-environment using Polycom SoundPoint IP 650 as extensions. When adding just one custom ringtone (~57KB) in a proper format (ML.wav: RIFF (little-endian) data, WAVE audio, ITU G.711 mu-law, mono 8000 Hz) the phone boots well. But after I have chosen the custom ringtone as my ringtone the phone works without problems until next reboot. I have to rename my custom ringtone for that it is not found on boottime, change the ringtone to a default one and reboot the phone to make it work again. My questions: 1. Where are configurations done with the Webserver of the phone stored? I guess must be somewhere in the tftpboot-dir on my asterisk/elastix server. But I can't recognize any file changes (compared timestamps) 2. Where can I find out how many space is left on my phone (some PDF-Guides from polycom say that about 160KB of custom ringtones or about 120 items in contact directory are fine for the phone - I have 7 items in my contact directory and just one (57KB also tried 38KB) custom ringtone 3. What else could be the problem for this behaviour? Thank your for helping me gettng started with asterisk Marco -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unable to receive calls (inbound)
Hello all. I have installed AsteriskNow 1.7.1 with all updates. I'm able to make outbound calls without any problem (the external calls are made via an analog line, and the receiver see the CID). However I'm unable to forward incoming calls to the destination I want. What happens is when I make an internal call I ear a bye. Bellow is the log of the internal call: -- Starting simple switch on 'DAHDI/1-1' == Starting DAHDI/1-1 at ,s,1 failed so falling back to exten 's' == Starting DAHDI/1-1 at ,s,1 still failed so falling back to context 'default' -- Executing [s@default:1] Playback(DAHDI/1-1, vm-goodbye) in new stack -- DAHDI/1-1 Playing 'vm-goodbye.ulaw' (language 'en') -- Executing [s@default:2] Macro(DAHDI/1-1, hangupcall) in new stack -- Executing [s@macro-hangupcall:1] GotoIf(DAHDI/1-1, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf(DAHDI/1-1, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf(DAHDI/1-1, 1?theend) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup(DAHDI/1-1, ) in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'DAHDI/1-1' in macro 'hangupcall' == Spawn extension (default, s, 2) exited non-zero on 'DAHDI/1-1' -- Executing [h@default:1] Macro(DAHDI/1-1, hangupcall,) in new stack -- Executing [s@macro-hangupcall:1] GotoIf(DAHDI/1-1, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf(DAHDI/1-1, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf(DAHDI/1-1, 1?theend) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup(DAHDI/1-1, ) in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'DAHDI/1-1' in macro 'hangupcall' == Spawn extension (default, h, 1) exited non-zero on 'DAHDI/1-1' -- Hungup 'DAHDI/1-1' Can you help me debug the problem? TIA, -vcf -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Internode weirdness
On 01/21/11 03:19, Tom Rymes wrote: On 01/19/2011 10:34 PM, Da Rock wrote: WARNING[988]: chan_sip.c:19069 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog '481cf0543743e6bb7006991d409ed3bc@150.101.178.33:5060'. Giving up. Have you tried disallowing re-invites? Sorry for the delay, but I've tried both yes and no- one of the first things I tried, but I get your reasoning. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mailing list question
Thank you Kevin. That's exactly the answer I was after. I'll see if I can get it 'stopped' at our server end. BTW - the reason I asked in here was so that everyone could see the answer and, hopefully, do the same. Thanks again! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: 20 January 2011 18:44 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Mailing list question On 01/20/2011 11:16 AM, Andrew Thomas wrote: Sorry Dannny - it didn't work :( I can only hope that someone at API has the answer. Thanks for trying :) API provides the physical services and bandwidth for the mailing lists, but does not operate them. If you go to the lists.digium.com site and choose the 'asterisk-users' mailing list, you can see there is a link to send a message to the list administrator(s)... which would probably be more effective than asking a question like this on the list itself :-) In any case, the answer is no... the lists are operated using Mailman software, and it essentially leaves the message bodies alone (although it does do scrubbing of attachments in some cases). Unless you want to include your signature as an attachment marked as something other than 'text', I don't believe there's any way to get the mailing list process to drop your signature block. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AST-2011-001: Stack buffer overflow in SIP channel driver
Could you please give me a feedback regarding this issue, I'm not sure of the
answer I got browsing the web
Thanks and Best Regards
Le mercredi 19 janvier 2011 09:14:55, Marc Leurent a écrit :
Good morning,
I have a simple question,
Is this problem would affect also an Asterisk 1.4.38 if Pedantic SIP
support: No in the Global Signalling Settings
For what I understood, no..
Or is it a simple way to postpone upgrade until next planned upgrade.
Best Regards
Le mardi 18 janvier 2011 17:35:31, Asterisk Security Team a écrit :
Asterisk Project Security Advisory - AST-2011-001
ProductAsterisk
SummaryStack buffer overflow in SIP channel driver
Nature of Advisory Exploitable Stack Buffer Overflow
SusceptibilityRemote Authenticated Sessions
Severity Moderate
Exploits KnownNo
Reported On January 11, 2011
Reported By Matthew Nicholson
Posted On January 18, 2011
Last Updated OnJanuary 18, 2011
Advisory Contact Matthew Nicholson mnichol...@digium.com
CVE Name
Description When forming an outgoing SIP request while in pedantic mode,
a
stack buffer can be made to overflow if supplied with
carefully crafted caller ID information. This vulnerability
also affects the URIENCODE dialplan function and in some
versions of asterisk, the AGI dialplan application as well.
The ast_uri_encode function does not properly respect the
size
of its output buffer and can write past the end of it when
encoding URIs.
Resolution The size of the output buffer passed to the ast_uri_encode
function is now properly respected.
In asterisk versions not containing the fix for this issue,
limiting strings originating from remote sources that will be
URI encoded to a length of 40 characters will protect against
this vulnerability.
exten = s,1,Set(CALLERID(num)=${CALLERID(num):0:40})
exten = s,n,Set(CALLERID(name)=${CALLERID(name):0:40})
exten = s,n,Dial(SIP/channel)
The CALLERID(num) and CALLERID(name) channel values, and any
strings passed to the URIENCODE dialplan function should be
limited in this manner.
Affected Versions
Product Release Series
Asterisk Open Source1.2.x All versions
Asterisk Open Source1.4.x All versions
Asterisk Open Source1.6.x All versions
Asterisk Open Source1.8.x All versions
Asterisk Business Edition C.x.x All versions
AsteriskNOW 1.5 All versions
s800i (Asterisk Appliance) 1.2.x All versions
Corrected In
Product Release
Asterisk Open Source 1.4.38.1, 1.4.39.1, 1.6.1.21, 1.6.2.15.1,
1.6.2.16.1, 1.8.1.2, 1.8.2.1
Asterisk Business Edition C.3.6.2
Patches
URL
Branch
http://downloads.asterisk.org/pub/security/AST-2011-001-1.4.diff1.4
http://downloads.asterisk.org/pub/security/AST-2011-001-1.6.1.diff
1.6.1
http://downloads.asterisk.org/pub/security/AST-2011-001-1.6.2.diff
1.6.2
Re: [asterisk-users] Does Asterisk support NI-1 (DMS 100) and NI-2 forT1s?
Zeeshan Zakaria Sent: Friday, January 21, 2011 6:11 AM For a client I am setting up a system which will use T1 PRI from Primus, who offer only NI-1 and NI-2 protocols for D-Channels. Previousely I have only used switchtypes euroISDN and National. Although the documentation says Asterisk does support NI-1 ans NI-2, but wanted to get your opinion if you have used these protocols on an Asterisk box and if there were any things to consider. If anybody has experience with Primus, it'll be more helpful. I'm using NI-2 with no problem (but haven't tried all features). You say you've used National-wouldn't that be NI-1 or NI-2? --Don -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Inbound routes
Hello all. I have installed AsteriskNow 1.7.1-64bits with freePBX. The system has 1 DAHDi card with 2 analog FXO ports (to pstn) and 1 FXS port connected to a FAX machine. I want the every call received on port FXO-2 to be redirected to the FAX machine. So, what I configured was that every call with DID 12345 (let's suppose that 12345 is the DID connected to the FXO-2)to be redirected to the extension associated to the FXS port. However, it seams that when the call is received, the trunk does not inform the DID as you can see by the log: -- Starting simple switch on 'DAHDI/2-1' -- Executing [s@from-pstn:1] NoOp(DAHDI/2-1, No DID or CID Match) in new stack -- Executing [s@from-pstn:2] Answer(DAHDI/2-1, ) in new stack -- Executing [s@from-pstn:3] Wait(DAHDI/2-1, 2) in new stack -- Executing [s@from-pstn:4] Playback(DAHDI/2-1, ss-noservice) in new stack -- DAHDI/2-1 Playing 'ss-noservice.ulaw' (language 'en') Is my assumption correct? How can I solve that? TIA, -vcf -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Does Asterisk support NI-1 (DMS 100) and NI-2 for T1s?
In article AANLkTi=dpw1oewcpvhbgf1g2ymgbjo4yaml0gbs_6...@mail.gmail.com, Zeeshan Zakaria zisha...@gmail.com wrote: For a client I am setting up a system which will use T1 PRI from Primus, who offer only NI-1 and NI-2 protocols for D-Channels. Previousely I have only used switchtypes euroISDN and National. Although the documentation says Asterisk does support NI-1 ans NI-2, but wanted to get your opinion if you have used these protocols on an Asterisk box and if there were any things to consider. If anybody has experience with Primus, it'll be more helpful. switchtype = national IS NI-2. See the description in zapata.conf: ; Switchtype: Only used for PRI. ; ; national: National ISDN 2 (default) ; dms100: Nortel DMS100 ; 4ess: ATT 4ESS ; 5ess: Lucent 5ESS ; euroisdn: EuroISDN ; ni1:Old National ISDN 1 ; qsig: Q.SIG So you use national for NI-2 and ni1 for the obsolete NI-1. Hope this helps. Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MOH and parking
I know that the 'fix' has just been applied (https://issues.asterisk.org/view.php?id=18262) - but why does it stop the moh only to start it again? This, also, seems to cause a CDR problem (see below). -- Executing [7000@chambers:1] Park(SIP/2000-0008, ) in new stack == Parked SIP/2000-0008 on 7001 (lot default). Will timeout back to extension [chambers] s, 1 in 60 seconds -- Added extension '7001' priority 1 to parkedcalls (0xb6fd1160) -- SIP/2000-0008 Playing 'digits/7.gsm' (language 'en') -- SIP/2000-0008 Playing 'digits/0.gsm' (language 'en') -- SIP/2000-0008 Playing 'digits/0.gsm' (language 'en') -- SIP/2000-0008 Playing 'digits/1.gsm' (language 'en') == Spawn extension (chambers, s, 1) exited non-zero on 'Parked/SIP/2000-0008ZOMBIE' -- Stopped music on hold on DAHDI/1-1 -- Started music on hold, class 'dv-ip', on DAHDI/1-1 [Jan 21 13:39:17] ERROR[22913]: cdr_addon_mysql.c:313 mysql_log: Failed to insert into database: (1062) Duplicate entry 'DV-IP-1295617064.8' for key 1 == Spawn extension (park-dial, SIP02000, 1) exited non-zero on 'SIP/2000-0008ZOMBIE' BTW 'DV-IP-1295617064.8' is the CDR entry for when the call first came in to the queue. So, it looks like it's trying to use the same unique ID again. If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inbound routes
On 01/21/2011 8:49 AM, Vitor Carlos Flausino wrote: The system has 1 DAHDi card with 2 analog FXO ports (to pstn) [snip] However, it seams that when the call is received, the trunk does not inform the DID This is because FXO ports do not support DID. You need to route the call based on the port it came in on, not based on a DID. Tom -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] res_fax
On 01/21/2011 08:37 PM, Tom Rymes wrote: On Jan 20, 2011, at 8:52 PM, Steve Underwood wrote: A comparison wouldn't be complete without mentioning Hylafax. Hylafax has a great infrastructure - tools for integrating with Windows clients, and so on. Neither spandsp or the Digium FAX code can match that for FAX termination. I think its biggest drawback is you either use it with iaxmodem for audio FAXing, or t38modem for T.38 FAXing. It can't smoothly integrate the two right now. As a longtime Hylafax user, I can confirm it's an excellent solution. I am somewhat surprised about the comment of being able to do audio or t.38, but not both. This is probably a little true and untrue at the same time, though I have never used t.38modem with Hylafax. Given the structure of the product, you could have HylaFAX connected to both an IAXModem and a T.38Modem at the same time (or 23 IAXModems, a 24-port T1/E1 PCI-card modem, and 7 t.38modems for that matter...). What it cannot do, is receive audio and t.38 on the same port, which is what I presume that Steve was referring to. This is really a limitation of IAXmodem and t.38modem, as one only handles audio, the other only handles t.38. In other words, you could route t.38 faxes to it on port 1 and audio faxes on port2, but you cannot have port 1 handle both types. Its easy to set up some t38modem channels and some iaxmodem channels for receiving FAXes. Transmit is more problematic. With this split config, you need to know in advance whether the particular number is accessible by T.38 or by audio. Most people won't. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] res_fax
On 01/21/2011 8:59 AM, Steve Underwood wrote: On 01/21/2011 08:37 PM, Tom Rymes wrote: On Jan 20, 2011, at 8:52 PM, Steve Underwood wrote: [snip] Its easy to set up some t38modem channels and some iaxmodem channels for receiving FAXes. Transmit is more problematic. With this split config, you need to know in advance whether the particular number is accessible by T.38 or by audio. Most people won't. Steve Good point. Perhaps you could route via chan_clairvoyant? Tom -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inbound routes
The system has 1 DAHDi card with 2 analog FXO ports (to pstn) [snip] However, it seams that when the call is received, the trunk does not inform the DID This is because FXO ports do not support DID. You need to route the call based on the port it came in on, not based on a DID. Ok... And how do I do that? I'm using freePBX. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inbound routes
This is how I have done it. In FreePBX, under the 'Setup' tab, choose 'Zap Channel DIDs'. Assign a DID (ex. 12345) to the channel(port). Then go to 'Inbound Routes' and create a route for the DID and set the destination to the appropriate extension. On 01/21/2011 08:21 AM, Vitor Carlos Flausino wrote: This is because FXO ports do not support DID. You need to route the call based on the port it came in on, not based on a DID. Ok... And how do I do that? I'm using freePBX. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Where are stored the CDR's?
Am 21.01.2011 12:21, schrieb Vitor Carlos Flausino: Hello all. Can you help me find where the CDR's are being stored? The result of cdr show status is: Call Detail Record (CDR) settings -- Logging: Enabled Mode: Simple Log unanswered calls: No * Registered Backends --- (none) /var/log/asterisk/cdr/ That directory does not exist. In fact there are two directories named cdr-csv and cdr-custom, but both are empty. Also, file /etc/asterisk/cdr.conf is empty and file cdr_mysql.conf contains: [global] hostname = localhost dbname=asteriskcdrdb password = fpbx user = freepbx userfield=1 However the database is empty. But... how can I find (which command inside asterisk), where the cdr are stored? Does the DB exist? If not, so try to create it: mysql -e CREATE DATABASE asteriskcdrdb mysql -e GRANT select, insert ON asteriskcdrd.* TO 'freepbx'@'localhost' IDENTIFIED BY 'fpbx' -Thorsten- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AST-2011-001: Stack buffer overflow in SIP channel driver
Thank you for the confirmation Best Regards, Le vendredi 21 janvier 2011 14:17:20, Kevin P. Fleming a écrit : On 01/21/2011 05:59 AM, Marc Leurent wrote: Could you please give me a feedback regarding this issue, I'm not sure of the answer I got browsing the web Thanks and Best Regards Le mercredi 19 janvier 2011 09:14:55, Marc Leurent a écrit : Good morning, I have a simple question, Is this problem would affect also an Asterisk 1.4.38 if Pedantic SIP support: No in the Global Signalling Settings For what I understood, no.. Or is it a simple way to postpone upgrade until next planned upgrade. The advisory clearly states that all Asterisk 1.4.x releases are affected when pedantic mode is enabled. Since you have pedantic mode disabled, your system is not vulnerable to this problem. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Does Asterisk support NI-1 (DMS 100) and NI-2 for T1s?
Yes, it does. Bell provides the same as well and it works with Asterisk. -Bruce On Fri, Jan 21, 2011 at 7:11 AM, Zeeshan Zakaria zisha...@gmail.com wrote: Hi list, For a client I am setting up a system which will use T1 PRI from Primus, who offer only NI-1 and NI-2 protocols for D-Channels. Previousely I have only used switchtypes euroISDN and National. Although the documentation says Asterisk does support NI-1 ans NI-2, but wanted to get your opinion if you have used these protocols on an Asterisk box and if there were any things to consider. If anybody has experience with Primus, it'll be more helpful. Thanks Zeeshan A Zakaria -- www.visionvoip.com www.ilovetovoip.com www.pbxforall.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Does Asterisk support NI-1 (DMS 100) and NI-2 for T1s?
Thanks a lot guys for your answers. I'll go ahead with NI-2. I didn't know its the same thing as National. Thanks again, Zeeshan A Zakaria -- www.visionvoip.com www.ilovetovoip.com www.pbxforall.com On 2011-01-21 10:36 AM, Bruce B bruceb...@gmail.com wrote: Yes, it does. Bell provides the same as well and it works with Asterisk. -Bruce On Fri, Jan 21, 2011 at 7:11 AM, Zeeshan Zakaria zisha...@gmail.com wrote: Hi list, For a client I am setting up a system which will use T1 PRI from Primus, who offer ... -- _ -- Bandwidth and Colo... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MOH and parking
On 11-01-21 08:52 AM, Andrew Thomas wrote: I know that the 'fix' has just been applied (https://issues.asterisk.org/view.php?id=18262) - but why does it stop the moh only to start it again? This, also, seems to cause a CDR problem (see below). After speaking with Shaun and Russell, this is likely related to some other part of code, and the fix that went in shouldn't have caused this issue. It's possible fixing this may have caused some other part of the code that was broken to be more prevalent though. Could you open an issue on bug tracker? Thanks! Leif. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Force Dahdi modules to load
People, I'm trying to force dahdi to load the modules I need to get spans working. I have two cards, an Digium TE210E (PCI-e) and a Yeastar TDM1600 FXO (PCI) Actually it is loading just the first of them, it is (wtc4xxp) for TE210E, but doesn't load the second module as specified at /etc/dahdi/modules: wct4xxp ystdm16xx I have to load it manually to get spans working. modprobe ystdm16xx I need to get around this, and force system to get both modules loaded at startup. I also have /etc/modprobe.d/dahdi.blacklist.conf : blacklist wct4xxp - Need this to load first blacklist ystdm16xx - Need this to load after blacklist ystdm8xx blacklist wcte12xp blacklist wct1xxp blacklist wcte11xp blacklist wctdm24xxp blacklist wcfxo blacklist wctdm blacklist wctc4xxp blacklist wcb4xxp blacklist netjet Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to asterisk t.38
Hi Bryant, The 1.4 box has two interfaces one with 202 ip and the other with 172 ip , the audio code has 172 ip and the ast 1.6 has only 172 ip. Any ideas ? Both the trunks have t.38 enabled on it. And the way we use fax is fax machine connected to ata which supports t.38 in both the ends. Thank You Amit Nepal Systems Administrator Phoenix Internet Phone: 602-385-0731 602-234-0917#112 http://www.phoenixinternet.net On 1/20/2011 4:11 PM, Bryant Zimmerman wrote: Amit Make sure that the trunk you have between the two servers has the t.38 enabled on it. Do you have any NAT between the two servers or are they on the same lan. We do the t.38 faxing between 1.4 and 1.6 asterisk boxes all of the time. Our audio codes gateway dumps into a 1.4 box and all faxes calls are then sent to either 1.6.x or 1.8.x boxes and then on to the final ata. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 *From*: Amit Nepal ami...@phoenixinternet.net *Sent*: Thursday, January 20, 2011 4:27 PM *To*: asterisk-users@lists.digium.com *Subject*: Re: [asterisk-users] Asterisk to asterisk t.38 Hi, I have an Audio code gateway between two asterisk servers. The audio code has PRI connected for PSTN. I can send faxes and receive faxes in ast 1.4 . Also I can send faxes for ast 1.6 to outside (PSTN) and receive faxes. The only problem I am having is sending/receiving between ast 1.4 and ast 1.6. ATA (T.38 capable) AST 1.6 AUDIO CODEAST 1.4ATA (t.38 Capable) Thank You Amit Nepal On 1/20/2011 1:56 PM, David Backeberg wrote: On Thu, Jan 20, 2011 at 3:14 PM, Amit Nepalami...@phoenixinternet.net wrote: I have a setup of asterisk 1.6 in one box and asteirsk 1.4 in another. I can send recieve faxes from both boxes fine to and from pstn. But the faxing between 1.6 and 1.4 extensions does fail. Any ideas please ? You don't say what's between the boxes as the medium over which the faxes are going. Try a fax between them without t.38 and see if it goes through. It might be a connection that is not reliable for any kind of faxing. That would not be an asterisk problem, it would be a faxing over a bad connection problem. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queues with ringinuse=yes
I'm setting up a queue for two independent operator phones that are capable of answering multiple calls at once. It's currently working with the following settings and Asterisk 1.4: queues.conf: [telefonistas] strategy=roundrobin ;strategy=leastrecent music=default timeout=60 retry=0 maxlen=0 wrapuptime=0 ringinuse=yes autofill=yes joinempty=yes member = SIP/8899 member = SIP/8898 extensions.conf: exten = _900[0-2],1,Answer() exten = _900[0-2],n,Queue(telefonistas,r,,,45) exten = _900[0-2],n,Dial(DAHDI/g3/9072) ;If no operator answers Both SIP phones are set up with call-limit=4 so each phone will receive a maximum of 4 calls before Asterisk considers the phone busy and don't sends calls to it anymore. The tricky part comes now: the customer asked me to load balance the phones, so the next incoming call should be sent to the phone with the least calls. For example: a) Phone1 has one call, Phone2 has two. Next incoming call should be routed to Phone1. b) Phone1 has two calls, Phone2 has zero. Next incoming call should be routed to Phone2. c) Both Phone1 and Phone2 have one call. Next incoming call can be routed to any of them. And so on. I have never done something similar before. I tried changing the queue strategy to leastrecent but that didn't solve the issue, because both Phone1 and Phone2 must also make calls in addition to answering them, and that should be taken in consideration. I'm not sure if such a thing is possible using Queue() at all. As a last resort I'll write a dialplan that will check how many calls each phone has and route appropriately, but I really would like to implement that using Queue() if possible. Any ideas? Thanks in advance. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Phone multi-registration
Hi, I'm working with a SIP phone (Thomson ST2030) that support a multi-registration feature. It works this way: - the phone has a main id - some feature keys can be configured to be tied to supplementary ids (with a specific id username and password) - when the phone boots, it will successively (try to) register with each id (main and supplementary ids). - every outgoing call uses the main id - every incoming call received for any id is handled. The trouble I'm having is I can configure either Asterisk or the SIP phone to have supplementary id successfully registered (403 Wrong Password reply though I'm certain the password I provided is correct). I could find an (impractical) workaround setting in sip.conf a supplementary id with a fixed IP address (and thus removing the need for registration). I'm suspecting a mistake in my config but I prefer to check here. Is it possible for a SIP to register twice to the same Asterisk server using 2 different ids ? Consulting this list archives gives mixed answers. Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Channel in an unkown state
Hello all. I have a dahdi card with 2 FXO and 1 FXS. I'm able to make calls without any problem. However, when I have an incoming call, I see the following message on the asterisk console: -- Starting simple switch on 'DAHDI/1-1' -- Executing [s@DID_trunk_1:1] ExecIf(DAHDI/1-1, 1?SetCallerPres(unavailable)) in new stack -- Auto fallthrough, channel 'DAHDI/1-1' status is 'UNKNOWN' -- Hungup 'DAHDI/1-1' Notice the status UNKNOWN. Can someone help me? Best regards, -vcf -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Phone multi-registration
On Fri, Jan 21, 2011 at 12:53 PM, Olivier oza_4...@yahoo.fr wrote: Hi, Is it possible for a SIP to register twice to the same Asterisk server using 2 different ids ? Consulting this list archives gives mixed answers. Yes, I do this with my Polycom 550, and I've done it with other phones before as well. Show us the CLI output and possibly a SIP Debug of the registration attempts. -- Thanks, --Warren Selby, dCAP http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] waitforsilence changed after upgrade to 1.6
Hi all. I have a customer who does some automatic messaging. Back when we were running Asterisk 1.4.x, they used waitforsilence after amd, to wait for an answering machine greeting to finish before leaving their message. We've upgraded to 1.6, and made no other changes and things don't work. What we're seeing it that a call will get hung up, waiting for silence, well after any sane greeting message would have finished. Up to an hour! I understand that waitforsilence take a 3rd parameter as a timeout, but all that seems to do for us is keep a channel from staying up for over, say, 5 minutes. We'd like to be able to start our message as soon as the greeting is done. Any suggestions? TIA, -- Take care and have fun, Mike Diehl. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ongoing problem with 1.8
At 10:33 AM 1/18/2011, you wrote: While that's a useful data point, it's not relevant to the problem. A significant portion of the SIP stack was re-implemented in 1.8, and Polycom phones are on the desktops of nearly every Asterisk developer. Since you aren't using a Polycom, the SIP stack on that device is implemented differently, causing possible incompatibilities. This is why the tcpdump will be helpful: to figure out what is different and why it doesn't work. Well, here it is. Please let me know if this helps or if there is anything else you might want. Ira [root@loretta asterisk]# tcpdump 'udp port 5060' tcpdump: verbose output suppressed, use -v or -vv for full protocol decode listening on eth0, link-type EN10MB (Ethernet), capture size 96 bytes 16:40:56.252047 IP sip.flowroute.com.sip 192.168.233.235.sip: SIP, length: 451 16:40:56.252710 IP 192.168.233.235.sip sip.flowroute.com.sip: SIP, length: 554 2 packets captured 2 packets received by filter 0 packets dropped by kernel [root@loretta asterisk]# tcpdump 'udp port 5060' tcpdump: verbose output suppressed, use -v or -vv for full protocol decode listening on eth0, link-type EN10MB (Ethernet), capture size 96 bytes 17:00:23.331450 IP sip.flowroute.com.sip 192.168.233.235.sip: SIP, length: 451 17:00:23.331569 IP sip.flowroute.com.sip 192.168.233.235.sip: SIP, length: 451 17:00:23.332978 IP 192.168.233.235.sip sip.flowroute.com.sip: SIP, length: 501 17:00:23.60 IP 192.168.233.235.sip sip.flowroute.com.sip: SIP, length: 501 17:00:27.676300 IP proxy.ideasip.com.sip 192.168.233.235.sip: SIP, length: 4 17:00:27.676451 IP proxy.ideasip.com.sip 192.168.233.235.sip: SIP, length: 4 17:00:29.919758 IP 192.168.233.240.sip 192.168.233.235.sip: SIP, length: 1241 17:00:29.920503 IP 192.168.233.235.sip 192.168.233.240.sip: SIP, length: 556 17:00:29.999893 IP 192.168.233.240.sip 192.168.233.235.sip: SIP, length: 441 17:00:30.016914 IP 192.168.233.240.sip 192.168.233.235.sip: SIP, length: 1419 17:00:30.018819 IP 192.168.233.235.sip 192.168.233.240.sip: SIP, length: 499 17:00:31.270141 IP 192.168.233.235.sip 192.168.233.240.sip: SIP, length: 787 17:00:31.474063 IP 192.168.233.240.sip 192.168.233.235.sip: SIP, length: 698 17:00:33.630719 IP 192.168.233.240.sip 192.168.233.235.sip: SIP, length: 802 17:00:33.631060 IP 192.168.233.235.sip 192.168.233.240.sip: SIP, length: 468 17:00:33.967906 IP 192.168.233.240.sip 192.168.233.235.sip: SIP, length: 802 17:00:33.968211 IP 192.168.233.235.sip 192.168.233.240.sip: SIP, length: 468 17:00:34.297189 IP 192.168.233.240.sip 192.168.233.235.sip: SIP, length: 802 17:00:34.297552 IP 192.168.233.235.sip 192.168.233.240.sip: SIP, length: 468 17:00:34.774236 IP 192.168.233.240.sip 192.168.233.235.sip: SIP, length: 802 17:00:34.774608 IP 192.168.233.235.sip 192.168.233.240.sip: SIP, length: 468 17:00:35.124020 IP 192.168.233.240.sip 192.168.233.235.sip: SIP, length: 802 17:00:35.124394 IP 192.168.233.235.sip 192.168.233.240.sip: SIP, length: 468 17:00:35.406606 IP 192.168.233.240.sip 192.168.233.235.sip: SIP, length: 802 17:00:35.406993 IP 192.168.233.235.sip 192.168.233.240.sip: SIP, length: 468 17:00:35.617205 IP 192.168.233.240.sip 192.168.233.235.sip: SIP, length: 802 17:00:35.617615 IP 192.168.233.235.sip 192.168.233.240.sip: SIP, length: 468 17:00:35.917571 IP 192.168.233.240.sip 192.168.233.235.sip: SIP, length: 802 17:00:35.917852 IP 192.168.233.235.sip 192.168.233.240.sip: SIP, length: 468 17:00:36.156349 IP 192.168.233.240.sip 192.168.233.235.sip: SIP, length: 802 17:00:36.15 IP 192.168.233.235.sip 192.168.233.240.sip: SIP, length: 468 17:00:36.525815 IP 192.168.233.240.sip 192.168.233.235.sip: SIP, length: 802 17:00:36.526105 IP 192.168.233.235.sip 192.168.233.240.sip: SIP, length: 468 17:00:36.886513 IP 192.168.233.240.sip 192.168.233.235.sip: SIP, length: 802 17:00:36.886773 IP 192.168.233.235.sip 192.168.233.240.sip: SIP, length: 468 17:00:41.504275 IP 192.168.233.235.sip 192.168.233.233.sip: SIP, length: 913 17:00:41.506291 IP 192.168.233.235.sip 192.168.233.237.sip: SIP, length: 915 17:00:41.508264 IP 192.168.233.235.sip 192.168.233.240.sip: SIP, length: 913 17:00:41.780812 IP 192.168.233.235.sip 192.168.233.233.sip: SIP, length: 913 17:00:41.781813 IP 192.168.233.235.sip 192.168.233.237.sip: SIP, length: 915 17:00:41.782807 IP 192.168.233.235.sip 192.168.233.240.sip: SIP, length: 913 17:00:42.332759 IP 192.168.233.235.sip 192.168.233.240.sip: SIP, length: 913 17:00:42.333765 IP 192.168.233.235.sip 192.168.233.237.sip: SIP, length: 915 17:00:42.334761 IP 192.168.233.235.sip 192.168.233.233.sip: SIP, length: 913 17:00:43.432667 IP 192.168.233.235.sip 192.168.233.240.sip: SIP, length: 913 17:00:43.437668 IP 192.168.233.235.sip 192.168.233.237.sip: SIP, length: 915 17:00:43.442660 IP 192.168.233.235.sip 192.168.233.233.sip: SIP, length: 913 17:00:45.632489 IP 192.168.233.235.sip 192.168.233.240.sip: SIP, length: 913 17:00:45.645484 IP
Re: [asterisk-users] spandsp download
Where can I get the latest stable version of spandsp. That will work with res_fax_spandsp.so. The link listed on the voip-info website is dead. Any other location for download? http://www.soft-switch.org/ Thanks Bryant Zimmerman -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
