Re: [asterisk-users] Friday Feb 29th Leap Year Special wih Aastra
On Fri, Feb 29, 2008 at 2:08 AM, Rob Hillis [EMAIL PROTECTED] wrote: If anyone has managed to compile and run Asterisk on a server from this particular era, I'd love to know about it. :) Reports on asterisk and SIP from Roman times are a little sketchy. However, in about 110 BC, Claudius Maximus wrote It's a good thing cellphones were inventer to make VoIP sound good! Let's hear about SIP/DECT from Aastra TODAY Feb 29th at 12 Noon EST. Details at http://x2z.eu /r What's the performance like? For that matter, what phones were available at the time? randulo wrote: Doesn't that depend on what planet you are currently on? Or like, if you were in ancient Rome or something, with a Cesarian calendar? ___ -- 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] Openser balancing Asterisk
Hi , i've a problem with the dispatcher module of Openser, for the load
balancing for asterisk
The schema of the network is this :
Firewall (public ip:
199.199.199.199:5060)
|
|
---
Pc1
pc2
Pc1: 192.168.0.1:5060 Asterisk
Pc2:192.168.0.3:5060 Asterisk
Pc1:192.168.0.2:5060 Openser
The ip public is natted with Openser
The dispatcher.list is:
1 sip:192.168.0.1:5060
1 sip:192.168.0.3:5060
Openser.cfg:
route{
if ((method==INVITE) ){
ds_select_dst(1,4); # 4 = round-robin
t_relay();
exit();
}
}
In asterisk for test there is an ivr ,and when I make a call from an other
site to this public ip, ast answer but after few seconds I see from the log
of the pc01 the the call hang but from the remote asterisk (the one where I
make the call) the call remain up.
The configuration of openser is correct ?
thanks
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Re: [asterisk-users] quickfix for building zaptel with 2.6.24?
On Thu, Feb 28, 2008 at 11:10:37AM -0600, Kevin P. Fleming wrote: Louis-David Mitterrand wrote: zenon:~# module-assistant -t build zaptel make[3]: Entering directory `/usr/src/linux-2.6.24.3' scripts/Makefile.build:46: *** CFLAGS was changed in /usr/src/modules/zaptel/Makefile. Fix it to use EXTRA_CFLAGS. Stop. Is there a quickfix out there? Yes, use Zaptel 1.4.9.1 or wait for the release of 1.4.10 later today or first thing tomorrow. If you decide to use 1.4.9.1, please note that if you are using analog cards with FXO modules, there is a known bug in DTMF generation that will affect your ability to dial out on those ports. That has been fixed in Subversion (see issue 11855 on bugs.digium.com) and will be in the next release. Thanks for your answer Kevin, but I need the debian'ized bristuff'ed version to be able to package and deploy it. I'll just patiently wait for Tzafrir (thanks for your work!) to release them for debian. Cheers, ___ -- 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] Cisco 7965g and asterisk
Hi,
We've just bought a new cisco 7965g and web are trying to connect it to
asterisk. I've bought smartnet and downloaded
cmterm-7945_7965-sccp.8-3-3SR2.exe
cmterm-7945_7965-sip.8-3-3SR2.zip
The zip file contains:
Archive: cmterm-7945_7965-sip.8-3-3SR2.zip
Length Date TimeName
2496963 11-05-07 10:06 apps45.8-3-3ES2.sbn
585536 11-05-07 10:06 cnu45.8-3-3ES2.sbn
2817314 11-05-07 10:07 cvm45sip.8-3-3ES2.sbn
326315 11-05-07 10:06 dsp45.8-3-3ES2.sbn
557452 11-05-07 10:07 jar45sip.8-3-3ES2.sbn
642 11-05-07 10:06 SIP45.8-3-3SR2S.loads
642 11-05-07 10:06 term45.default.loads
642 11-05-07 10:06 term65.default.loads
---
6785506 8 files
How can i install the sip firmware?
Thanks
Nuno Fernandes
Overflow
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[asterisk-users] basic installation
Hi I have just joined this group and i need to know what it take to have asterisk setup. What are the requirement. I ahve to submit a project for my college on VoIP and thouhg Astrerisk would be a great platform . Any help will really be great !!! -- Regards Agnello Dsouza www.linux-vashi.blogspot.com www.bible-study-india.blogspot.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] which phones to use ??
I am following the tutorial give at this link http://chayden.net/Asterisk/SeUpAsteriskAtHome.htm but it does not mention the phones that i need to use could i use any USB phone !!! ??? thansk !! -- Regards Agnello Dsouza www.linux-vashi.blogspot.com www.bible-study-india.blogspot.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 queue agent problem
Dear I have queue setup arround 10 agent setup now what happend when call inter in queue and queue transfer call available extension but suppose extension A call to extension B ( ineternal sip call ) that time anycall come into queue and suppose queue transfer call on A extension so it got busy caz its talking to B ext. so my call got hangup ...so how to my queue detect which extension got busy or free is there any function my queue detect channel STATUS ??? Thing is that my queue only know about agent phone status when queue transfer call to agent...right if 2 agent talk to each othere that time my queue not come in that part so how does my queue understand A and B extesion are busy ??? PGP Signature-- Satish Patel mobile:- +91-9818875535 http://www.linuxbug.org - Looking for last minute shopping deals? Find them fast with Yahoo! Search.___ -- 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] Cisco 7965g and asterisk
On Fri, 2008-02-29 at 11:09 +, Nuno Fernandes wrote: Hi, We've just bought a new cisco 7965g and web are trying to connect it to asterisk. I've bought smartnet and downloaded [snip] How can i install the sip firmware? You need to setup a tftp server, put the 8 sip firmware files and the configuration files in the tftp server directory so the Cisco phone can pick them up when it boots. The Cisco phone can be very picky about the configuration files. With the slightest error the phone will refuse to boot so make sure you have got it all right. I don't have configuration files for the 7965 so ask the company where you bought the phone or google around. If you can't find them for the 7965 please note that the 7940/7960 configuration files will not work for a 7965. Maybe the 7941/7961 configuration files will. I'm not sure. If you are mainly using Asterisk (SIP) then I recommend you buy Polycom, Aastra or Snom phones next time. The Polycom phones have the best sound quality and imho are the best SIP phones you can buy. Regards, Patrick ___ -- 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] basic installation
On Fri, Feb 29, 2008 at 12:33 PM, Agnello George [EMAIL PROTECTED] wrote: I have just joined this group and i need to know what it take to have asterisk setup. What are the requirement. I ahve to submit a project for my college on VoIP and thouhg Astrerisk would be a great platform . These two articles, though old, are an excellent intro to asterisk: http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html The Book TFOT http://www.oreilly.com/catalog/asterisk/ ___ -- 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] which phones to use ??
On Fri, Feb 29, 2008 at 1:12 PM, Agnello George [EMAIL PROTECTED] wrote: but it does not mention the phones that i need to use could i use any USB phone !!! ??? I would recommend you start by using free softphones like X-Lite, Gizmo project, Zoiper. Then, when you're ready, choose a hardphone by price and quality needed between Grandstream, Sipura, Polycom and Cisco not to mention Snom or Aastra that have a lot of models as well. ___ -- 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] Problems with removing zaptel
Hi, Will have a look. many thanks. On 2008-02-29 at 11:35 Paul Hales wrote: /etc/modprobe/blacklistor similar PaulH On Fri, 2008-02-29 at 00:30 +0100, Christian wrote: Hi all, Using the latest test version of Debian but when I have done modprobe -r and removed a few of the zaptel modules some of them cannot be removed. The other module is in use. Also if I reboot my system they're all loaded again. Any thoughts? Many thanks, Christian ___ -- 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] Problems with removing zaptel
Hi, Many thanks for that info, will give it a try. All the best, Christian On 2008-02-29 at 09:09 Tzafrir Cohen wrote: On Fri, Feb 29, 2008 at 12:30:49AM +0100, Christian wrote: Hi all, Using the latest test version of Debian but when I have done modprobe -r and removed a few of the zaptel modules some of them cannot be removed. The other module is in use. Also if I reboot my system they're all loaded again. Any thoughts? modprobe -r does not recursively remove modules. Try: /etc/init.d/zaptel unload #if using the init.d script from the deb /etc/init.d/zaptel stop # if using the init.d script from the tarball (The reason for the difference: there is no point in unlading a module on system shutdown. Only serves to increase the crash potential) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/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
Re: [asterisk-users] TDM400P dialout problem
Anthony Messina wrote: i'm looking forward to 1.4.9.2, but am also concerned about http://bugs.digium.com/view.php?id=12099 as i saw this error with 1.4.9 and 1.4.9.1 on both platforms. The messages in bug 12099 are *not* errors, they are annoyances only. The latest SVN branch 1.4 code of Asterisk will no longer generate them, and once my battery_alarms branch has been merged into Zaptel 1.4 (scheduled to be part of the 1.4.10 release) then Zaptel will stop generating spurious battery alarm events. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- 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 queue agent problem
Hi Satish You would want to investigate Local channels on Asterisk for this. Garth Garth van Sittert BSc (Physics Computer Science) - Main: 08600 BITCO Phone: +27 (0)11 875 6900 Fax:+27 (0)11 875 6901 Mobile: +27 (0)83 791 6662 Email: [EMAIL PROTECTED] MSN:[EMAIL PROTECTED] Web:www.bitco.co.za satish patel wrote: Dear I have queue setup arround 10 agent setup now what happend when call inter in queue and queue transfer call available extension but suppose extension A call to extension B ( ineternal sip call ) that time anycall come into queue and suppose queue transfer call on A extension so it got busy caz its talking to B ext. so my call got hangup ...so how to my queue detect which extension got busy or free is there any function my queue detect channel STATUS ??? Thing is that my queue only know about agent phone status when queue transfer call to agent...right if 2 agent talk to each othere that time my queue not come in that part so how does my queue understand A and B extesion are busy ??? PGP Signature-- Satish Patel mobile:- +91-9818875535 http://www.linuxbug.org Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://us.rd.yahoo.com/evt=51734/*http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- 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] when we try to add CURL code to file channel.c we get an error - undefined reference to curl_easy_init
Hi all, When I try to add CURL code to file channel.c we get an error - undefined reference to curl_easy_init. I've added #include curl/curl.h so the code compiles fine. this error is generated by the linker, even though func_curl.c is compiled and linked with no errors My asterisk machine have curl and curl-devel 7.12 installed. Asterisk version i am using is 1.4.17. Any help will be appriciated. Thanks Regards Prashant Sharma ___ -- 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] Pattern matching....
Tony Mountifield wrote: In article [EMAIL PROTECTED], Eric Wieling [EMAIL PROTECTED] wrote: No that will not work. You would want three exten = lines, one for each area code. And if you have a lot of common dialplan that you don't want to duplicate between the three extension patterns, put the common stuff up at a higher priority and use Goto to get there: exten = _404NXX,1,Goto(200) exten = _770NXX,1,Goto(200) exten = _678NXX,1,Goto(200) exten = _NXXNXX,200,NoOp(Start of common instructions) exten = _NXXNXX,n,etc Actually you can do this a lot more simply without using Goto (which might mess with your CDR): exten = _404NXX,1,NoOp exten = _770NXX,1,NoOp exten = _678NXX,1,NoOp exten = _NXXNXX,2,NoOp(Start of common instructions) exten = _NXXNXX,n,etc Since there is an implicit 'Goto' from priority 1 to priority 2 anyway, you might as well take advantage of it :-) -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- 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 as useragent registered using 2 accounts
Thanx for the tip. It has erased the problem i was having using authentication but another problem has arised. i am now able to call with only one user from AST1 to AST2. If i dial using the other user, my AST2 shows the following warning and responds with a 403 forbidden sip response: *WARNING[13520]: chan_sip.c:8117 check_auth: username mismatch, have adf, digest has abc* Any solutions to this problem? On Wed, Feb 27, 2008 at 4:36 PM, Igor A. Goncharovsky [EMAIL PROTECTED] wrote: Rizwan Hisham wrote: I am having a strange problem. I am using my asterisk server AST1 to register with another asterisk server AST2 using 2 accounts (2 register commands in sip.conf). I have made 2 local users in AST1, and configured my dialplan in such a way that both local accounts on AST1 use different trunks to send the call to AST2 server. These 2 different trunks are for 2 accounts i have registered on AST1. (skiped) How can i make asterisk realize it? You must enable authentication of INVITE that AST1 send to AST2. Now you have no authentication of incoming INVITE and AST2 make decision about used account based only on IP address of caller peer. Changing insecure=port,invite to insecure=port should help. -- Best regards, Igor A. Goncharovsky ___ -- 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 -- Best Regards Rizwan Hisham ___ -- 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] load balancing and high availability
I am evaluating the best way to make a high avail and load balanced system. I have two identical asterisk servers. Most clients are SIP phones. The only special hardware I have on both systems (they are identical) is: 1 E1 PRI card and 1 4-port BRI card. I have 8 ISDN lines so 4 go to each pbx server. I have 2 PRI lines that connect to an Alcatel PBX so each asterisk pbx has 1 PRI connection (routed the same way of course). I need to implement an active-active cluster of 2 servers. I'm new to Heartbeat and I've read this: http://www.ultramonkey.org/3/topologies/ha-lb-eg.html Could I setup Asterisk with this topology? Would I just need to have 2 identical servers? Would call routing/SIP registrations/internal astdb be handled correctly (ie. as if it were a single server)? Thanks for your input. Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs ___ -- 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] when we try to add CURL code to file channel.c we get an error - undefined reference to curl_easy_init
On Friday 29 February 2008 08:10:40 Prashant Sharma wrote: When I try to add CURL code to file channel.c we get an error - undefined reference to curl_easy_init. I've added #include curl/curl.h so the code compiles fine. this error is generated by the linker, even though func_curl.c is compiled and linked with no errors My asterisk machine have curl and curl-devel 7.12 installed. Asterisk version i am using is 1.4.17. Let's start with, why are you adding curl code to channel.c? -- 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] TDM400P dialout problem
Anthony Messina wrote: with 1.4.7.1, i had no problems with either x86_64 or i386. with 1.4.8, i386 worked, but x86_64 did not. with 1.4.9 and 1.4.9.1, neither worked. i use the rpms from atrpms.net for fedora 7 i'm looking forward to 1.4.9.2, but am also concerned about http://bugs.digium.com/view.php?id=12099 as i saw this error with 1.4.9 and 1.4.9.1 on both platforms. unfortunately, due to my work schedule, i did not have time to debug the differences between the platforms. There are no differences between the platforms for this bug. The bug was caused by the use of uninitialized memory, which could contain any (random) data. As such, each time you ran ztcfg, it would cause (or clear up) the bug for any given DTMF digit that your system might generate. Using Zaptel 1.4.8, 1.4.9 or 1.4.9.1 meant that if you got correct DTMF generation for all digits then you were lucky; the platform had nothing to do with it :-) -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- 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] Gtalk with asterisk
Hi, I have been working with Asterisk for the ivr functionalities in the past. I am interested to implement the Jabber - Gtalk in asterisk. For which i installed the iksemel but this didnt help me out. I couldnt find the res_jabber.so file any where in the asterisk source directory. Still when i run the command make menuselect the channel driver chan_gtalk shows xxx (dependencies not met). How can i register gtalk with asterisk. If you can provide me with some basic details i can carry forward. Thanks and appreciate your response. Regards, Naveen.Palani “Quinnox, a global IT services company prides itself on its SEI-CMM Level 5, ISO‑9001:2000 assessed delivery processes and provides solutions in areas of E-Business, ERP, Application Management Services, and EAI to customers in BFSI, Manufacturing, Retail, Telecom and Healthcare sector, powered by our Global Delivery Model.” This e-mail and any attached files are confidential, proprietary, and may also be legally privileged information, and are intended solely for the use of the individual or entity to whom they are addressed. If you are not the intended recipient of this e-mail, please send it back to the person who sent it to you and delete the e-mail and any attached files and destroy any copies of it; you may call us immediately at + 91 22 2829 0100 or email us at [EMAIL PROTECTED] Quinnox Consultancy Services and/or any of its sister companies owns no responsibility for the views presented in the e-mail and any attached files unless the sender mentions so, with due authority of Quinnox Consultancy Services. Unauthorized reading, reproduction, publication, use, dissemination, forwarding, printing or copying of this e-mail and its attachments is prohibited. We have checked this message for any known viruses; however we decline any liability, in case of any damage caused by a non-detected virus. For more details about our company, visit http://www.Quinnox.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] Listening to Allison voicemail prompt on SIP phone causes [pop] sounds.
Ok so I'm not going crazy then. I filed a bug report. http://bugs.digium.com/view.php?id=12093 -Original Message- From: Trevor Peirce [mailto:[EMAIL PROTECTED] Sent: Wednesday, February 27, 2008 6:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Listening to Allison voicemail prompt on SIP phone causes [pop] sounds. shadowym wrote: A bit hard to describe. Using a SIP hardphone I log into my voicemail at which point Allison says you have x messages.. There are various other prompts that exhibit the same problem but that is one easy to explain and reproduce one. The problem is there is a slight 'pop' sound usually during the first syllable of each word. So the prompt sounds like y[pop]ou h[pop]ave t[pop]wo me[pop]ssages. If I dial *43 to do an echo test and Allison says you are about to enter an echo test. it's not there so it's only in certain modes this happens. Yes, this is something that has bothered me since I first started working with asterisk 1.2 way back when. It sounds to me like it's an artifact of appending multiple sound files together as it occurs at the beginning of each prompt that is played, or each digit when reading back caller id. I too see this with gsm, ulaw, and the new slin files. I know it happens with 1.2 and 1.4 on Sipura/Linksys ATAs. I just listened to the prompts on my Aastra 9112i and the pop is there too but not nearly as apparent as on the ATAs. I've got no idea where to even start trying to solve something like this, but I just wanted to respond that you're not the only one being bothered by it. Best regards, Trevor Peirce -- Real CNAM data for incoming Caller ID @ www.cnam.info ___ -- 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] Detecting SIT (Special Information Tone) on outbound calls
For reference of SIT please check http://en.wikipedia.org/wiki/Special_information_tone Regards, Sanjay. - Original Message - From: sanjay rajdev [EMAIL PROTECTED] To: asterisk-users asterisk-users@lists.digium.com Sent: Friday, February 29, 2008 8:35:08 PM (GMT+0530) Asia/Calcutta Subject: [asterisk-users] Detecting SIT (Special Information Tone) on outbound calls Is there a way to detect SIT (Special Information Tone) when making an outbound call. Regards, Sanjay. ___ -- 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] bugs.digium.com
Tracker seems to be down. -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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] Page app, Polycom IP 601, 60 SIP peers, Interesting Issue
Hi All, I have a pretty standard Asterisk PBX setup with 60 SIP Peers, mostly Polycom 501's and a receptionist phone, Polycom IP 601 with 3 attached sidecars and Buddy Watch enabled monitoring all other SIP phones. The problem occurs when a group (all SIP peers) Page is called. Not always but sometimes when the Page is executed, the IP 601 will become unreachable from Asterisk. So when the receptionist hangs up the page, the BYE doesn't get back to Asterisk to release all the Page channels so they stay open. I have to restart Asterisk to release all the open SIP Channels. What I think is happening is when all the SIP peers are paged, Asterisk sends 60 hint notifications to the IP 601 and the phone is overloaded and can't respond to SIP POKE or process the BYE message back to Asterisk properly. I'm wondering if I upgrade to a new IP 650 with a faster processor, will this eliminate the issue? Has anyone experienced this or have ideas for resolution or further troubleshooting? Thanks. JR -- JR Richardson Engineering for the Masses ___ -- 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] bugs.digium.com
On Fri, Feb 29, 2008 at 3:38 PM, Doug Lytle [EMAIL PROTECTED] wrote: Tracker seems to be down. Can't be! Mark once told me, The bug tracker is never on vacation! after I chided him on how much he worked when on vacation. ___ -- 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] Detecting SIT (Special Information Tone) on outbound calls
Is there a way to detect SIT (Special Information Tone) when making an outbound call. Regards, Sanjay. ___ -- 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] Listening to Allison voicemail prompt on SIP phone causes [pop] sounds.
shadowym wrote: Ok so I'm not going crazy then. The jury is still out on that issue! John Novack I filed a bug report. http://bugs.digium.com/view.php?id=12093 -Original Message- From: Trevor Peirce [mailto:[EMAIL PROTECTED] Sent: Wednesday, February 27, 2008 6:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Listening to Allison voicemail prompt on SIP phone causes [pop] sounds. shadowym wrote: A bit hard to describe. Using a SIP hardphone I log into my voicemail at which point Allison says you have x messages.. There are various other prompts that exhibit the same problem but that is one easy to explain and reproduce one. The problem is there is a slight 'pop' sound usually during the first syllable of each word. So the prompt sounds like y[pop]ou h[pop]ave t[pop]wo me[pop]ssages. If I dial *43 to do an echo test and Allison says you are about to enter an echo test. it's not there so it's only in certain modes this happens. Yes, this is something that has bothered me since I first started working with asterisk 1.2 way back when. It sounds to me like it's an artifact of appending multiple sound files together as it occurs at the beginning of each prompt that is played, or each digit when reading back caller id. I too see this with gsm, ulaw, and the new slin files. I know it happens with 1.2 and 1.4 on Sipura/Linksys ATAs. I just listened to the prompts on my Aastra 9112i and the pop is there too but not nearly as apparent as on the ATAs. I've got no idea where to even start trying to solve something like this, but I just wanted to respond that you're not the only one being bothered by it. Best regards, Trevor Peirce -- 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] Request for testing: New wctdm24xxp and wcte12xp drivers.
Hi All, This is a request for testing for users of Asterisk 1.4 or 1.6 with any of the following Digium VoiceBus based cards: TDM2400P, AEX2400, AEX800, TDM800P, TDM410, TE120P, TE121, and/or TE122. From a practical standpoint, this branch should allow these boards to work in more systems / configurations where IRQ misses caused problems in the past. The updated wctdm24xxp and wcte12xp drivers are based on Zaptel 1.4 and work by dynamically increasing the latency they add to the data stream in 1ms increments until it does not detect any additional IRQ misses. For example, if your TDM800P card was sharing an interrupt with your disk controller and you were experiencing problems, this new driver could, at the cost of a few extra milliseconds of latency, allow the two boards to peacefully coexist on the same IRQ. Please feel free to checkout the code on any test servers you may have and let me know any results. Both the good and the bad are welcome. To get the code: svn co http://svn.digium.com/svn/zaptel/team/sruffell/voicebus Many thanks, -- Shaun Ruffell Linux Kernel Developer Digium, 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] Page app, Polycom IP 601, 60 SIP peers, Interesting Issue
Oh yes! This has been killing us for about a year. We've had several conference calls with my phone vendor and Polycom and it's still not fixed (or even determined why it is happening). Polycom keeps saying, upgrade to the next version of the firmware. We upgrade, still a problem. (again, for over a year!) In my case, the Polycom 601 actually reboots when we page! When it comes back up, I have a phantom meetme on the Asterisk system and none of the sidecar lights are correct. Sometimes, they simply stop updating completely. Just FYI, go to the CLI and type meetme. You'll get the conference ID and the number of users. Then, type meetme kick confID 01 Using, of course, the conference ID. The 01 is the user that initiated the meetme. So, when you kick 01, the rest go away politely! This keeps us from having to restart Asterisk. We are on Bootrom 3.2.3.0002 and SIP 2.2.0.0047 as of yesterday and we STILL have the problem. Our setup is one Polycom 601 and 25 Polycom 501s that are being paged. The 601 is powered by PoE with 2 sidecars, so Polycom wants us to put an actual Power Supply on the phone - thinking the voltage is dropping and causing the reboot. I don't buy that, but we are putting one on next Monday. We'll see. Our next plan is to get a 650 and see if it can handle the traffic. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of JR Richardson Sent: Friday, February 29, 2008 9:17 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Page app, Polycom IP 601, 60 SIP peers, Interesting Issue Hi All, I have a pretty standard Asterisk PBX setup with 60 SIP Peers, mostly Polycom 501's and a receptionist phone, Polycom IP 601 with 3 attached sidecars and Buddy Watch enabled monitoring all other SIP phones. The problem occurs when a group (all SIP peers) Page is called. Not always but sometimes when the Page is executed, the IP 601 will become unreachable from Asterisk. So when the receptionist hangs up the page, the BYE doesn't get back to Asterisk to release all the Page channels so they stay open. I have to restart Asterisk to release all the open SIP Channels. What I think is happening is when all the SIP peers are paged, Asterisk sends 60 hint notifications to the IP 601 and the phone is overloaded and can't respond to SIP POKE or process the BYE message back to Asterisk properly. I'm wondering if I upgrade to a new IP 650 with a faster processor, will this eliminate the issue? Has anyone experienced this or have ideas for resolution or further troubleshooting? Thanks. JR -- JR Richardson Engineering for the Masses ___ -- 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] Gtalk with asterisk
Hi Naveen, For which i installed the iksemel but this didnt help me out. I couldnt find the res_jabber.so file any where in the asterisk source directory. Still when i run the command make menuselect the channel driver chan_gtalk shows xxx (dependencies not met). How can i register gtalk with asterisk. If you can provide me with some basic details i can carry forward. Either the iksemel library has not been properly installed on your system, or Asterisk did not detect it. The following link may help you, as it includes hints to troubleshoot iksemel + GnuTLS installations : http://www.voip-info.org/wiki/view/Asterisk+Google+Talk However, feel free to open a bug report if you've made sure you have a properly installed iksemel stack. Note that as of Asterisk 1.6, GnuTLS was replaced by OpenSSL which is now used by Asterisk as the encrypting protocol for iksemel. -- Philippe Sultan ___ -- 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] IAX2's JB and DTMF
We've moved within the last two months to Asterisk 1.4.x All remote facilities are connected via highspeed (9mbit) connections (Over OpenVPN) to a central Asterisk box, acting as a voice router, that funnels all calls into our Avaya Definity G3R via PRI. When corporate employees visit the remote facilities and try to call the G3R's voice mail system(Audix), DTMF is not recognized unless you enter the digits in VERY slowly. I've been doing testing today and found that if I either set up a SIP trunk (With jbenable=yes) or disable the jitter buffer on the iax trunk, I can get a reliable reading from the G3R. Is there anything that can be tweaked to allow IAX with JB enabled to increase the reliability of the DTMF generation? I'm having problems enabling SIP trunks on the production systems that I'm working on, so at the moment can't use them. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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] SPA3102 registration problem
Quoting Mandeep Singh Bhabha [EMAIL PROTECTED]: what i did to configure SPA3102 is ... My problem is that normal SPA3102 configurations just don't seem to work. I can't even get the FXS port to register. I'm beginning to suspect that my unit is defective. Here's why: If I configure the FXS port to register with my Asterisk server using the most basic sip.conf configuration *without* a password, then it does actually register. The only problem is the address it gives: bitis*CLI sip show peers Name/username HostDyn Nat ACL Port Status 8000/8000 127.0.0.1D 5060 OK (1 ms) That's right: instead of 192.168.1.8, it's telling Asterisk that it's available on the loopback address! I'll bet this is why it's not able to register using a password. I got the same results with three different firmware versions. Still, it's an unusual way for a device to be broken. Could there be another reason for this behavior? Otherwise I'll just have to try to explain it to the vendor and perhaps to Linksys tech support and ask for my money back. Cheers, Jaap ___ -- 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] Skewed RTP timestamps in SIP calls on Asterisk 1.4.18
Last week I migrated some of our servers to Asterisk 1.4.18 and we started seeing audio drops of several seconds during SIP calls. After investigating it we noticed that Asterisk was increasing the RTP timestamps abnormally during a conversation. I'm including a text file with a subset of the data collected by Wireshark that shows the problem (I have the complete packet capture if anybody needs it to analyze it). The Asterisk server is the one whose IP address ends in .38. If you look at the packet with the number 14910 (seq 23369) you'll see that the next packet from Asterisk (14919, seq 23370) increases the RTP timestamp from 77120 to 2280582632. We've tried enabling and disabling internal timing and the jitter buffer, but it made no difference whatsoever. I also added the patch present in ticket #10355 to Asterisk 1.4.18, but it didn't help. Has anybody else experienced a problem like this one? 14898 52.678422 63.215.27.5566.150.122.38 RTP PT=ITU-T G.711 PCMU, SSRC=0x352D1216, Seq=22216, Time=1043051951 14899 52.678576 66.150.122.38 63.215.27.55RTP PT=ITU-T G.711 PCMU, SSRC=0x404D77E4, Seq=23368, Time=76960 14909 52.698326 63.215.27.5566.150.122.38 RTP PT=ITU-T G.711 PCMU, SSRC=0x352D1216, Seq=22217, Time=1043052111 14910 52.699321 66.150.122.38 63.215.27.55RTP PT=ITU-T G.711 PCMU, SSRC=0x404D77E4, Seq=23369, Time=77120 14917 52.718417 63.215.27.5566.150.122.38 RTP PT=ITU-T G.711 PCMU, SSRC=0x352D1216, Seq=22218, Time=1043052271 14919 52.720938 66.150.122.38 63.215.27.55RTP PT=ITU-T G.711 PCMU, SSRC=0x404D77E4, Seq=23370, Time=2280582632 14921 52.721029 66.150.122.38 63.215.27.55RTP PT=ITU-T G.711 PCMU, SSRC=0x404D77E4, Seq=23371, Time=2280582792 14922 52.721052 66.150.122.38 63.215.27.55RTP PT=ITU-T G.711 PCMU, SSRC=0x404D77E4, Seq=23372, Time=2280582952 ___ -- 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] Polycom IP600 + PC share same switch port with VLAN
As far as I can tell, with Polycom phones you cannot do what you're asking (which is for the PC and the phone to be in the same VLAN while the PC is connected to the phone). I don't know how they handle it when the voice frames are untagged, but they definitely won't pass tagged voice frames to the PC port: http://knowledgebase.polycom.com/KanisaPlatform/Publishing/616/12526_f.SAL_PUBLIC_1_2.html Your switch port is configured for untagged frames on a single VLAN (that's what access mode is). Polycom phones need voice and data to be on separate VLANs in order for you to use the PC port. Since Polycom phones apparently don't support the Cisco Voice VLAN feature, you need to configure the port as a trunk port, which will allow you to send multiple VLANs to the phone. The phone will take frames tagged for your designated voice VLAN, and will pass the rest on to the PC port. For example: interface FastEthernet2/0/1 switchport trunk encapsulation dot1q switchport trunk native vlan XXX switchport trunk allowed vlan XXX,100 switchport mode trunk spanning-tree portfast trunk Replace XXX with whatever your PC VLAN is. Setting XXX as the native VLAN for this port will cause frames in that VLAN to be untagged for that port, which is what your PC probably expects. If it happens to be 1, then it's the native VLAN by default. The last command may or may not be available, depending on your version of IOS. If it isn't, portfast just won't work and you're just stuck with STP negotiation anytime the port bounces. -James On Fri, Feb 29, 2008 at 12:13 AM, Lee, John (Sydney) [EMAIL PROTECTED] wrote: Thanks very much for the quick response. However, switchport voice vlan.. I thought is only valid for CISCO phones and I am using Polycom and thus it would not work. Furthermore, I have already tried switchport voice vlan... before I emailed to the list. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bob G Sent: Friday, 29 February 2008 5:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom IP600 + PC share same switch port with VLAN You can paste and copy innterface FastEthernet2/0/1 switchport access vlan 20 switchport mode access switchport voice vlan 120 srr-queue bandwidth share 10 10 60 20 srr-queue bandwidth shape 10 0 0 0 mls qos trust device cisco-phone mls qos trust cos auto qos voip cisco-phone spanning-tree portfast - Original Message - From: Lee, John (Sydney) To: asterisk-users@lists.digium.com Subject: [asterisk-users] Polycom IP600 + PC share same switch port with VLAN Date: Fri, 29 Feb 2008 16:39:24 +1100 Hi all, I have been googling and testing without any luck and would appreciate any guidance from anyone. A port has already been configured on the CISCO switch with the following: interface FastEthernet2/0/1 description VOIP VLAN 100 switchport access vlan 100 switchport mode access duplex full speed 100 I plugged the phone into the port and everything worked as far as VOIP is concerned. Then I plug a PC into the PC port of the Polycom phone with the hope that I only need one port to support 2 devices. (I wanted the VOIP phone to use VLAN 100 and PC just the native VLAN) PROBLEM: However, I found that I could not get the PC (using DHCP) to get an IP address at all. It seems to be that the traffic from the PC is also tagged as VLAN 100 as well. I was told by others that there is a setting on the Polycom phone which allows the traffic of the PC, under this type of settings, to go native. Can anyone please help? 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 -- Want an e-mail address like mine? Get a free e-mail account today at www.mail.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 ___ -- 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] which phones to use ??
For your own sanity's sake, steer as far away from Grandstream as possible. The firmware is appalling and isn't improving a great deal. They make great steps in one area while another gets worse and worse. randulo wrote: On Fri, Feb 29, 2008 at 1:12 PM, Agnello George [EMAIL PROTECTED] wrote: but it does not mention the phones that i need to use could i use any USB phone !!! ??? I would recommend you start by using free softphones like X-Lite, Gizmo project, Zoiper. Then, when you're ready, choose a hardphone by price and quality needed between Grandstream, Sipura, Polycom and Cisco not to mention Snom or Aastra that have a lot of models as well. ___ -- 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] Polycom IP600 + PC share same switch port with VLAN
Have you set the VLAN tag on the phone? CP Lee, John (Sydney) wrote: Hi all, I have been googling and testing without any luck and would appreciate any guidance from anyone. A port has already been configured on the CISCO switch with the following: interface FastEthernet2/0/1 description VOIP VLAN 100 switchport access vlan 100 switchport mode access duplex full speed 100 I plugged the phone into the port and everything worked as far as VOIP is concerned. Then I plug a PC into the PC port of the Polycom phone with the hope that I only need one port to support 2 devices. (I wanted the VOIP phone to use VLAN 100 and PC just the native VLAN) PROBLEM: However, I found that I could not get the PC (using DHCP) to get an IP address at all. It seems to be that the traffic from the PC is also tagged as VLAN 100 as well. I was told by others that there is a setting on the Polycom phone which allows the traffic of the PC, under this type of settings, to go native. Can anyone please help? 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 ___ -- 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] callpark feature in ABE?
Hi All - Anyone know if the callpark feature is in ABE? Is there a comprehensive list of the differences between ABE and the open source version? I've only seen a bullet-point chart which has no real detail. Thanks, Noah ___ -- 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] which phones to use ??
On Sat, Mar 1, 2008 at 1:07 AM, Rob Hillis [EMAIL PROTECTED] wrote: For your own sanity's sake, steer as far away from Grandstream as possible. The firmware is appalling and isn't improving a great deal. They make great steps in one area while another gets worse and worse. I know a lot of people feel thzat way about GS, but many have had good luck with them. They're cheap, there's no getting around that, and they make a decent starter phone. Ironically, on the VoIP Users Conference yesterday, a person using a BT102 had better sound than many other callers in the past. I would recommend that if anyone has a firmware version that works, don't ever change it 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] Help asterisk connectivity with MS SQL
hi all I want to connect my asterisk system with a MS sql .I have done it with MYSQL but i want to connect it with MS SQL.I have tried a lot but not getting anything.Can anybody help me on this.' Thanks in advance Rahul ___ -- 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] which phones to use ??
On Sat, 1 Mar 2008, randulo wrote: On Sat, Mar 1, 2008 at 1:07 AM, Rob Hillis [EMAIL PROTECTED] wrote: For your own sanity's sake, steer as far away from Grandstream as possible. The firmware is appalling and isn't improving a great deal. They make great steps in one area while another gets worse and worse. I know a lot of people feel thzat way about GS, but many have had good luck with them. They're cheap, there's no getting around that, and they make a decent starter phone. Ironically, on the VoIP Users Conference yesterday, a person using a BT102 had better sound than many other callers in the past. I would recommend that if anyone has a firmware version that works, don't ever change it though :) Not just firmware, but hardware... The latest 1.1.5.15 Seems to be good - on newer phones, but if you've got older phones, then 1.1.1.14 is the one for you ... I've deployed many 2000's now - currently have a pair of 1200's on trial with a customer and finding them to be very good so-far. Build quality is a bit fisher price, but for the features vs. price, I've not found anything to beat it yet... 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
