Re: [asterisk-users] Asterisk mem leak behavior?
Really, what I would do is to set up a daily restart point when there is no or very little activity, something like running nightly: asterisk -rx stop when convenient and then having the monitoring script restart it immediately. Do you need to unload the zaptel modules as well or is restarting Asterisk enough? l. PS. one of the things I like less about asterisk is that every time you have a different piece of iron connected to a PBX, you end up patching zaptel and recompiling. so all ot of time you find yourself wondering if the patch will be successful with the latest zaptel version or not :-( On Tue, 29 Jan 2008 07:24:49 +0100, Mark Greene [EMAIL PROTECTED] wrote: So here is my setup. Hardware: Intel P3 1.2 Ghz 1 GB RAM 36 GB Drives Mirrored Software: CentOS 5 2.6.18 Kernel Asterisk 1.4.14 Zaptel 1.4.7 (redfone) LIbpri 1.4.2 -- Loway Research - Home of QueueMetrics http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maybe a little OT---USB Handset
Thanks for the replies. I wonder if I could use the Yealink phone and write a connector to Asterisk with the IAX client on Sourceforge and make the handset look like an iaxphone? Or maybe there is some other easier solution? All I need is to have the ability to go off hook/on hook, pass DTMF, and voice obviously :-) JohnM John, I've done something similar. I had a simple commandline IAX phone running on a low powered computer (actually an NSLU2 - but it could be any linux machine), driving a USB audio card. I didn't need a keyboard, as it auto answered incoming calls. If you are interested, contact me off list, and perhaps I can help. Tim. ___ -- 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] Do Asterisk requires audio codec to be installed?
Hi, Can you please tell me whether Asterisk requires any audio or video codec to be installed separately or it supports itself? Thanking you, Preeta Please do not print this email unless it is absolutely necessary. Spread environmental awareness. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.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] Dialogic card
On Jan 28, 2008 11:12 PM, Edgar Guadamuz [EMAIL PROTECTED] wrote: Hi list, Anyone knows where I can get information about configuring a Dialogic card to run with Asterisk?? The model I have is D/120JCT-LS. Somebody told me that I had to buy the driver, but I don't know if this is true and if so, who, how and how much... I was under the impression that only ABE supports Dialogic boards. I thought I saw that in passing so I could be totally wrong. Thanks, Steve Totaro ___ -- 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] Change Default Voicemail Message
Hi Dan, Sorry to bring a thread back from the dead, but you might find the following interesting. Is there an easy way to achieve this with a computer generated voice? We do not wish to manually record the messages if possible, in the interests of a consistent message across all voicemail boxes. What would be the easiest way to do this? It was almost as if the guys at Nerd Vittles knew exactly what you needed and wrote an article about it... :) http://nerdvittles.com/index.php?p=202 Headline is: Allison’s Text-to-Speech Trifecta: Cepstral, Asterisk 1.4, and FreePBX 2.3 Opening line: If you've longed for a text-to-speech Asterisk toolkit that sounds just like the default Allison prompts that ship with Asterisk 1.4, then today is your lucky day. Havn't tried it myself but hopefully it comes close to what you need it for. Regards, Chris Bennett ___ -- 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's DANGEROUS Transfer CDR's
It's not Asterisk, it's SIP. Transfer takes the signaling off the Asterisk box. In features.conf, replace blind transfer with a call to a macro. Then redo your dialplan with the 'g' option on inward dial commands. When the called party uses the transfer command, your macro should read the digits to call and then store them in the db, a unique global, or GROUP () variable. Then it should hang up. This will cause the calling leg to exit the dial command to the next priority which should be a check of the variable. If digits are present, use the dial command to call them at your provider. No fuss, no muss. You should make sure the peer entry for the outbound side includes canreinvite=yes so only the signaling remains on your box and the media is invited off. You should also ignore calls to your macro that hit from the inbound call leg. Just return immediately and neither side will ever know the inbound call leg left for a moment. Sent from my iPhone On Jan 28, 2008, at 11:56 PM, Grey Man [EMAIL PROTECTED] wrote: Hi All, PLEASE READ if you depend on Asterisk CDR's and support transfers. Apologies for the shout but I'm desperate to get others to agree Asterisk has a big problem with the CDR's that are generated for transfers. I can understand why not too many people are interested as transfers are complicated and messy. However for those of us having to support transfers and depending on Asterisk CDR's for our billing we are in a sticky predicament! For anyone using Asterisk in a provider environment unaware of any problem I urge you to do a simple blind transfer on your system and check your CDR's. Most Asterisk based providers I tested are blocking transfers but I did find some other providers out there missing billable call legs! My goal is to try and get acknowledgement that there is a serious problem here that warrants a re-think about how Asterisk CDR's are generated. In an effort to succinctly encapsulate the problem I've produced the call and CDR flows below. Hopefully they make sense but if not I'm more than happy to elaborate and share my test results (the flows below won't be legibile without a mono spaced font, copy and pasting into notepad will make them readable). Blind Transfer (1.2 and 1.4): Time CallsCDRs | Dest | Dur(s) | |---|| T0 -| Alice -- * -- Bob | || | | || Tt -| Carol -- * -- Bob -| Bob | Tt | | | || Te -| End -| Carol | Te | Attended Transfer (1.2): Time CallsCDRs | Dest | Dur(s) | |---|| T0 -| Alice -- * -- Bob | || | | || T1 -| Alice -- * -- Carol | || | | || Tt -| Carol -- * -- Bob -| Bob | Tt | | | Carol | Tt - T1| | | s | Tt | | | || Te -| End -| s | Te | Attended Transfer (1.4): Time CallsCDRs | Dest | Dur(s) | |---|| T0 -| Alice -- * -- Bob | || | | || T1 -| Alice -- * -- Carol | || | | || Tt -| Carol -- * -- Bob -| || | | || Te -| End -| Bob | Te | | Bob | Te - T1| To put it another way here are some examples of how Asterisk systems and transfers can be exploited. 1. Place a call to a mobile you plan on having a lengthy call to. As soon as the call is establised blind transfer it to a low or free cost destination. You will only be billed for the mobile call up to the time it takes you to do the transfer the remainder of the call will be billed at the low cost or free destination. 2. With Asterisk 1.4 place a call to two billable destinations and then transfer them together. You'll only be billed for each destination up until the time it takes you to transfer. 3. With Asterisk 1.2 place a call to a low cost or free destination. Then place a call to an expensive destination and do an attended transfer. You'll only be billed for the expensive destination up unitl the time it takes to do the transfer. I have opened a bug on the issue but I suspect without input from others having the same problem it will fade away. http://bugs.digium.com/view.php?id=11849 From my point of view the design solution to this problem would be as simple as changing
Re: [asterisk-users] PRI Alarms, Comes Back, But Asterisk Won't Touch It!
On 29 Jan 2008, at 11:08, George Pajari wrote: Here is the scenario: Asterisk 1.14.13; zaptel 1.4.6; Digium TE120P (same problem with various previous versions; same problem with different TE120P cards). The customer has a partial (10 B-Channel) PRI that when it is busy (eight or more B channels in use), tends to fail as shown below... [Jan 26 23:00:31] ERROR[31893] chan_zap.c: Write to 28 failed: Unknown error 500 [Jan 26 23:00:31] ERROR[31893] chan_zap.c: Short write: 0/15 (Unknown error 500) [Jan 26 23:00:31] WARNING[31893] chan_zap.c: Detected alarm on channel 1: Red Alarm we then see every channel fail with a write error followed by a Red Alarm. Then [Jan 26 23:00:34] VERBOSE[7646] logger.c: == Primary D-Channel on span 1 down [Jan 26 23:00:34] WARNING[7646] chan_zap.c: No D-channels available! Using Primary channel 24 as D-channel anyway! [Jan 26 23:00:36] NOTICE[7647] chan_zap.c: Alarm cleared on channel 1 (alarm cleared messages for all channels deleted) [Jan 26 23:00:36] NOTICE[7646] chan_zap.c: PRI got event: No more alarm (5) on Primary D-channel of span 1 Yet even with all alarms cleared, a pri show span 1 command shows Status: Provisioned, Down, Active. It appears that Asterisk is not recovering from the errors. Restarting Asterisk will not bring the PRI back up -- that requires the zaptel drivers to be unloaded and reloaded. Why is this happening and what can be done about it? - What are the Telco techs seeing? I my experience (in the UK at least) it is always worth having a chat with them to see how a PRI problem looks from their end. At a guess (and an un-informed one at that) I'd say that they are waiting to see the line cycle completely before attempting to recover from the red alarm. Reloading the zaptel drivers does something pretty low level (drops then brings back the carrier?) which they see and respond to. We had one (failover) circuit which was configured to only recover when physically unplugged! Tim. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PRI Alarms, Comes Back, But Asterisk Won't Touch It!
Here is the scenario: Asterisk 1.14.13; zaptel 1.4.6; Digium TE120P (same problem with various previous versions; same problem with different TE120P cards). The customer has a partial (10 B-Channel) PRI that when it is busy (eight or more B channels in use), tends to fail as shown below... [Jan 26 23:00:31] ERROR[31893] chan_zap.c: Write to 28 failed: Unknown error 500 [Jan 26 23:00:31] ERROR[31893] chan_zap.c: Short write: 0/15 (Unknown error 500) [Jan 26 23:00:31] WARNING[31893] chan_zap.c: Detected alarm on channel 1: Red Alarm we then see every channel fail with a write error followed by a Red Alarm. Then [Jan 26 23:00:34] VERBOSE[7646] logger.c: == Primary D-Channel on span 1 down [Jan 26 23:00:34] WARNING[7646] chan_zap.c: No D-channels available! Using Primary channel 24 as D-channel anyway! [Jan 26 23:00:36] NOTICE[7647] chan_zap.c: Alarm cleared on channel 1 (alarm cleared messages for all channels deleted) [Jan 26 23:00:36] NOTICE[7646] chan_zap.c: PRI got event: No more alarm (5) on Primary D-channel of span 1 Yet even with all alarms cleared, a pri show span 1 command shows Status: Provisioned, Down, Active. It appears that Asterisk is not recovering from the errors. Restarting Asterisk will not bring the PRI back up -- that requires the zaptel drivers to be unloaded and reloaded. Why is this happening and what can be done about it? -- George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102 www.netvoice.ca www.ip-centrex.ca www.ip-pbx.ca www.vpas.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) ___ -- 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] OT: Call for beta testers (well... perhaps late Alpha).
SIP wrote: We've just launched the beta of a free service which is, really, still only JUST out of the alpha stages. http://www.voipmagnet.com The basic idea is this: it's an opt-in directory focused on VoIP contact info (with elements of social networking and privacy control). Again, the service is very rough, but we'd like input from the VoIP community. There are a good many things that are likely buggy, broken, or not yet implemented, but we feel that's what a beta is for. If you have any questions/concerns/issues/feedback/abuse, feel free to send it to me directly via email or post it to the list. Some things we know will be changed soon: -ability to add multiple VoIP accounts with each login (this is in the DB, but the interface elements are there yet -- we didn't like the way they looked first pass 'round) -ability to invite friends to join up so you can share contact info with friend groups without making everything (or anything) public -search optimisations (searching is functional, but a bit rough. We're looking for any and all input there, of course) We've a whole plate load of ideas for what to cram into the service, but the intent is to keep it free and provider-agnostic, but still maintain a centralised location where anyone can go and look up friends or coworkers to find out what VoIP service they're using and how to get in touch. We're open to any and all suggestions to what to add/change/fix to make this a service out of which the community will get some use. We're getting some good feedback on the service, and we've made some optimisations to the search functionality, but we're still looking for some aggressive testing. If anyone has a second or two to check things out, we'd greatly appreciate it. ___ -- 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 mem leak behavior?
Mark, I thought I would chime in here on your problem. Oddly, I have having the same issue with a PRI with similar symptoms. The odd part is that I have never had an issue like this with a asterisk PRI setup. My setup is a PRI with a Sangoma card with the exact same issue with 1.4.14. After a few days we are unable to communicate with the PRI, The D-channel goes offline as well but the physical circuit stays up with no alarms. It doesn't give one a comfort level with uptime. I had also re-compiled the asterisk 1.4.14 along with zaptel and libpri sources and it still failed. I have since updated to the latest asterisk, zaptel and libpri .17 with the hopes that it will be fixed. I thought perhaps the card may have had an issue but now I am beginning to wonder. Kevin _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Greene Sent: Tuesday, January 29, 2008 9:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk mem leak behavior? I've tried exiting the CLI in hopes that my being in there, though it wouldn't make any sense, was keeping it from restarting. No luck. I've already setup a cron script to restart asterisk at night when there is no traffic going over it. But I hate to just treat the symptoms. I want to solve the problem. It's hard to sleep knowing there is a ghost in one of my machines. It only takes restarting asterisk, nothing else, including zaptel. Once asterisk restarts it's ready to go. I can't make heads or tails of it. There are no PRI errors when all this going on either. Debug shows nothing by usual comm chatter between the system and C/O. - Mark No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.19.15/1248 - Release Date: 1/28/2008 9:32 PM ___ -- 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 mem leak behavior?
Kevin, After upgrading to the latest build of everything have you seen the problem anymore? What's your hardware and software configs? Maybe we can find a similarity in our systems. - Mark On Jan 29, 2008 9:53 AM, Kevin Kiely [EMAIL PROTECTED] wrote: Mark, I thought I would chime in here on your problem. Oddly, I have having the same issue with a PRI with similar symptoms. The odd part is that I have never had an issue like this with a asterisk PRI setup. My setup is a PRI with a Sangoma card with the exact same issue with 1.4.14. After a few days we are unable to communicate with the PRI, The D-channel goes offline as well but the physical circuit stays up with no alarms. It doesn't give one a comfort level with uptime. I had also re-compiled the asterisk 1.4.14 along with zaptel and libpri sources and it still failed. I have since updated to the latest asterisk, zaptel and libpri .17 with the hopes that it will be fixed. I thought perhaps the card may have had an issue but now I am beginning to wonder. Kevin -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Mark Greene *Sent:* Tuesday, January 29, 2008 9:49 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Asterisk mem leak behavior? I've tried exiting the CLI in hopes that my being in there, though it wouldn't make any sense, was keeping it from restarting. No luck. I've already setup a cron script to restart asterisk at night when there is no traffic going over it. But I hate to just treat the symptoms. I want to solve the problem. It's hard to sleep knowing there is a ghost in one of my machines. It only takes restarting asterisk, nothing else, including zaptel. Once asterisk restarts it's ready to go. I can't make heads or tails of it. There are no PRI errors when all this going on either. Debug shows nothing by usual comm chatter between the system and C/O. - Mark No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.19.15/1248 - Release Date: 1/28/2008 9:32 PM ___ -- 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] softmodems bank for ast.
Hi. We need a full featured modem bank 20+ to attend data calls. IAXmodem only supports fax protocols because spandsp only support fax protocols. The idea is to do a IAX wrapper like IAXmodem but with a full featured (but propietary) softmodem library like PCTEL or linuxant. I hate astribank+hardmodems solution. Some body think that is possible? Sorry for my horrible English. ;) Bye. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.18-rc2 Now Available
What is the command to obtain official release notes? On Tue, 29 Jan 2008, The Asterisk Development Team wrote: Asterisk 1.4.18-rc2 is now available. One of the developers made a change to chan_sip that they wanted to get in to this release. A few other bug fixes were added, as well. This release candidate is published for anyone that is interested in helping to test it for a couple of days before it is officially released. To download the release candidate, use the following svn command: $ svn co http://svn.digium.com/svn/asterisk/tags/1.4.18 asterisk-1.4.18-rc2 If you would like it in tarball format, use the following commands: $ svn export http://svn.digium.com/svn/asterisk/tags/1.4.18 asterisk-1.4.18-rc2 $ tar -czvf asterisk-1.4.18-rc2.tar.gz asterisk-1.4.18-rc2/ 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ -- 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 1.4.18-rc2 Now Available
Asterisk 1.4.18-rc2 is now available. One of the developers made a change to chan_sip that they wanted to get in to this release. A few other bug fixes were added, as well. This release candidate is published for anyone that is interested in helping to test it for a couple of days before it is officially released. To download the release candidate, use the following svn command: $ svn co http://svn.digium.com/svn/asterisk/tags/1.4.18 asterisk-1.4.18-rc2 If you would like it in tarball format, use the following commands: $ svn export http://svn.digium.com/svn/asterisk/tags/1.4.18 asterisk-1.4.18-rc2 $ tar -czvf asterisk-1.4.18-rc2.tar.gz asterisk-1.4.18-rc2/ Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SET with pipe symbol
On Tuesday 29 January 2008 08:32:44 Arjan Kroon | Mobillion wrote:
I want to place a pipe symbol in a variable by using the command Set
I tried the following code:
Set(M_CHANNELVAR=${UNIQUEID}|${CALLERID(number))
When I call to my applicatie I see the following output in my CLI :
Ignoring entry '612345678' with no = (and not last 'options'
entry)
(in my test call ${CALLERID(number) = 061234578)
I tried to escape the pipe symbol by using \ (backslash)
With the same result
Also I tried to place the variable between single or double quotes, but
with the same result.
Does anybody now how place a pipe symbol in variable.
You can't, in 1.4. This is by design. We have removed this restriction in
1.6. As a workaround, in 1.4, use the NoOp instruction with the SET dialplan
function, i.e.
NoOp(${SET(M_CHANNELVAR=${UNIQUEID}|${CALLERID(number))})
--
Tilghman
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[asterisk-users] codec_g729a.so problem...
Recently with Asterisk 1.4.17 I've been running into some stability issues. I started looking through my logs, and I found this: [Jan 29 09:41:45] WARNING[13132]: loader.c:620 inspect_module: Module 'codec_g729a.so' was not compiled against a recent version of Asterisk and may cause instability. I'm using the newest version of codec_g729a.so from the Digium website (v33). I've tried using the 686, 586, and 386 versions (my platform is 32 bit). Is there a version that has been properly compiled for 1.4.17? I never had this problem with previous versions of Asterisk (1.4.15 and below), and I can't seem to find any information on this. ___ -- 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] test please ignore
Ian wrote: Just testing to see if my emails to this mailing list gets through. Tried posting a question, but it failed Thanks Ian ___ -- 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 I as well have been having rotten luck lately with the mailing list. Replies to questions get axed. New questions get axed. I mailed the list admin, and never got a reply... for all I know, the mail I sent got axed. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial agent channel - busy
show agents: 6001 (First Agent) available at '6001' (musiconhold is 'default') 6002 (Test Agent) available at '6002' (musiconhold is 'default') 2 agents configured [2 online , 0 offline] __ show queues: testQueue2 has 0 calls (max unlimited) in 'ringall' strategy (10s holdtime), W:0, C:1, A:0, SL:100.0% within 60s Members: Agent/6002 (Not in use) has taken 1 calls (last was 88511 secs ago) No Callers testQueuehas 0 calls (max unlimited) in 'ringall' strategy (11s holdtime), W:0, C:1, A:0, SL:100.0% within 60s Members: Agent/6001 (Not in use) has taken 1 calls (last was 101522 secs ago) No Callers __ Thomas On Tuesday 29 January 2008 02:13:10 Paul Hales wrote: What does 'show agents' give you? 'show queues' would be useful too. PaulH -- Thomas Kenner ___ -- 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] transcoder
Cisco routers with DSPs as ip2ip gw will do it if you want to spend a few bucks On Jan 29, 2008, at 2:36 PM, Khaled Chehab [EMAIL PROTECTED] wrote: Dears Any one knows a standalone voip transcoder software name,not an ip pbx. What I want is to transcode the incoming sip calls from g711 to g723 or ilbc or g729 . and forward it to a media gateway .. Regards Khaled chehab * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- 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] Asterisk's DANGEROUS Transfer CDR's
On 1/29/08, Richard Revels [EMAIL PROTECTED] wrote: It's not Asterisk, it's SIP. Transfer takes the signaling off the Asterisk box. In features.conf, replace blind transfer with a call to a macro. Then redo your dialplan with the 'g' option on inward dial commands. When the called party uses the transfer command, your macro should read the digits to call and then store them in the db, a unique global, or GROUP () variable. Then it should hang up. This will cause the calling leg to exit the dial command to the next priority which should be a check of the variable. If digits are present, use the dial command to call them at your provider. No fuss, no muss. You should make sure the peer entry for the outbound side includes canreinvite=yes so only the signaling remains on your box and the media is invited off. You should also ignore calls to your macro that hit from the inbound call leg. Just return immediately and neither side will ever know the inbound call leg left for a moment. Sent from my iPhone On Jan 28, 2008, at 11:56 PM, Grey Man [EMAIL PROTECTED] wrote: Hi All, PLEASE READ if you depend on Asterisk CDR's and support transfers. Apologies for the shout but I'm desperate to get others to agree Asterisk has a big problem with the CDR's that are generated for transfers. I can understand why not too many people are interested as transfers are complicated and messy. However for those of us having to support transfers and depending on Asterisk CDR's for our billing we are in a sticky predicament! For anyone using Asterisk in a provider environment unaware of any problem I urge you to do a simple blind transfer on your system and check your CDR's. Most Asterisk based providers I tested are blocking transfers but I did find some other providers out there missing billable call legs! My goal is to try and get acknowledgement that there is a serious problem here that warrants a re-think about how Asterisk CDR's are generated. In an effort to succinctly encapsulate the problem I've produced the call and CDR flows below. Hopefully they make sense but if not I'm more than happy to elaborate and share my test results (the flows below won't be legibile without a mono spaced font, copy and pasting into notepad will make them readable). Blind Transfer (1.2 and 1.4): Time CallsCDRs | Dest | Dur(s) | |---|| T0 -| Alice -- * -- Bob | || | | || Tt -| Carol -- * -- Bob -| Bob | Tt | | | || Te -| End -| Carol | Te | Attended Transfer (1.2): Time CallsCDRs | Dest | Dur(s) | |---|| T0 -| Alice -- * -- Bob | || | | || T1 -| Alice -- * -- Carol | || | | || Tt -| Carol -- * -- Bob -| Bob | Tt | | | Carol | Tt - T1| | | s | Tt | | | || Te -| End -| s | Te | Attended Transfer (1.4): Time CallsCDRs | Dest | Dur(s) | |---|| T0 -| Alice -- * -- Bob | || | | || T1 -| Alice -- * -- Carol | || | | || Tt -| Carol -- * -- Bob -| || | | || Te -| End -| Bob | Te | | Bob | Te - T1| To put it another way here are some examples of how Asterisk systems and transfers can be exploited. 1. Place a call to a mobile you plan on having a lengthy call to. As soon as the call is establised blind transfer it to a low or free cost destination. You will only be billed for the mobile call up to the time it takes you to do the transfer the remainder of the call will be billed at the low cost or free destination. 2. With Asterisk 1.4 place a call to two billable destinations and then transfer them together. You'll only be billed for each destination up until the time it takes you to transfer. 3. With Asterisk 1.2 place a call to a low cost or free destination. Then place a call to an expensive destination and do an attended transfer. You'll only be billed for the expensive destination up unitl the time it takes to do the transfer. I have opened a bug on the issue but I suspect without input from others having the same problem it will fade away.
Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's
- Original Message From: Richard Revels [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, 29 January, 2008 12:21:16 PM Subject: Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's It's not Asterisk, it's SIP. Transfer takes the signaling off the Asterisk box. In features.conf, replace blind transfer with a call to a macro. Then redo your dialplan with the 'g' option on inward dial commands. When the called party uses the transfer command, your macro should read the digits to call and then store them in the db, a unique global, or GROUP () variable. Then it should hang up. This will cause the calling leg to exit the dial command to the next priority which should be a check of the variable. If digits are present, use the dial command to call them at your provider. No fuss, no muss. You should make sure the peer entry for the outbound side includes canreinvite=yes so only the signaling remains on your box and the media is invited off. You should also ignore calls to your macro that hit from the inbound call leg. Just return immediately and neither side will ever know the inbound call leg left for a moment. Sent from my iPhone Hi Richard, I'm not actually sure we're talking about the same thing here. It's not transfers I have a problem with it's the CDRs the transferred calls end up generating. In this case I am the provider and transfers through our Asterisk servers work fine it's just that we can't properly bill for them. Regards, Greyman. Make the switch to the world's best email. Get the new Yahoo!7 Mail now. www.yahoo7.com.au/worldsbestemail ___ -- 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] Do Asterisk requires audio codec to be installed?
Asterisk supports a whole bunch of codecs in the regular install - ulaw, alaw, gsm,ilbc being the more popular ones. A common paid codec is g729 - avbl at digium.com -rajeev On 1/29/08, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi, Can you please tell me whether Asterisk requires any audio or video codec to be installed separately or it supports itself? Thanking you, Preeta Please do not print this email unless it is absolutely necessary. Spread environmental awareness. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.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] transcoder
Dears Any one knows a standalone voip transcoder software name,not an ip pbx. What I want is to transcode the incoming sip calls from g711 to g723 or ilbc or g729 . and forward it to a media gateway .. Regards Khaled chehab * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- 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] SET with pipe symbol
Hi,
I want to place a pipe symbol in a variable by using the command Set
I tried the following code:
Set(M_CHANNELVAR=${UNIQUEID}|${CALLERID(number))
When I call to my applicatie I see the following output in my CLI :
Ignoring entry '612345678' with no = (and not last 'options'
entry)
(in my test call ${CALLERID(number) = 061234578)
I tried to escape the pipe symbol by using \ (backslash)
With the same result
Also I tried to place the variable between single or double quotes, but
with the same result.
Does anybody now how place a pipe symbol in variable.
Kind Regards,
image001.gif___
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[asterisk-users] test please ignore
Just testing to see if my emails to this mailing list gets through. Tried posting a question, but it failed Thanks Ian ___ -- 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 mem leak behavior?
I've tried exiting the CLI in hopes that my being in there, though it wouldn't make any sense, was keeping it from restarting. No luck. I've already setup a cron script to restart asterisk at night when there is no traffic going over it. But I hate to just treat the symptoms. I want to solve the problem. It's hard to sleep knowing there is a ghost in one of my machines. It only takes restarting asterisk, nothing else, including zaptel. Once asterisk restarts it's ready to go. I can't make heads or tails of it. There are no PRI errors when all this going on either. Debug shows nothing by usual comm chatter between the system and C/O. - Mark ___ -- 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 mem leak behavior?
Kevin, After upgrading to the latest build of everything have you seen the problem anymore? Don't know yet, waiting for it to break ( not a good feeling as you know) What's your hardware and software configs? Maybe we can find a similarity in our systems. It's a dell poweredge with a basic config, 23b + 1 d, Sangoma a101, hw-d-channel. What HW are you using and are you using any SPANDSP? - Mark ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.18-rc2 Now Available
Alex Balashov wrote: What is the command to obtain official release notes? On Tue, 29 Jan 2008, The Asterisk Development Team wrote: Asterisk 1.4.18-rc2 is now available. One of the developers made a change to chan_sip that they wanted to get in to this release. A few other bug fixes were added, as well. This release candidate is published for anyone that is interested in helping to test it for a couple of days before it is officially released. To download the release candidate, use the following svn command: $ svn co http://svn.digium.com/svn/asterisk/tags/1.4.18 asterisk-1.4.18-rc2 If you would like it in tarball format, use the following commands: $ svn export http://svn.digium.com/svn/asterisk/tags/1.4.18 asterisk-1.4.18-rc2 $ tar -czvf asterisk-1.4.18-rc2.tar.gz asterisk-1.4.18-rc2/ Thanks! The ChangeLog file will be in the svn checkout. If you only want to download the ChangeLog you can use this: svn export http://svn.digium.com/svn/asterisk/tags/1.4.18/ChangeLog -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] When does Asterisk REFER?
I was wondering under what conditions Asterisk will hand off a call to another switch. I'm trying to verify that my local PSTN's Coppercom switch operates correctly... and wanted to know how to get a call REFER'd to another end-point. Thanks, -Philip ___ -- 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] POE draw on Aastra 480i
Allen Casteran wrote: Anyone know what the POE draw is for the Aastra 480i phones? We have switches that will do 15 watts on 12 ports but only do 7.7 watts on all 24 ports. A Cisco 3560 switch will do 15.6 watts on all 24 ports. Just trying to find out if we need that much power. Drew wrote: According to Aastra tech support, 5 watts (peak) per 480i. We are testing five phones running on a Linksys SRW208P that will only support full 15W on up to 4 of 8 ports. I can power up the switch while all phones are connected without any issues. I would expect your lower power switch will provide ample power. But, PoE class does not matter? Did you plug five Aastra phones? I'm suspicious about how that scenario worked, I mean, as far as i know Aastra phones should register as a zero PoE class, that means it would reserve up to 12.94 watts no matter how many watts uses. So, my guess here is even if the phone use only 5 watts, the switch already reserved 12.94 watts for it. I would love to see what happens if you plug a sixth phone or figure out if you used an Aastra phone. Can you tell us what model/brand you used? Dimensioning PoE devices over capable switches has been a new issue which involves many factors like those described before. Regards, PD. Sorry about the original thread break off, I've been unable to find the original one. -- Octavio H. Ruiz Cervera Neocenter, SA. de CV. http://www.neocenter.com/ Soluciones para Centros de Contacto y Telefonía IP Tel.: (+52 55) 8590-9000 Ext. 9016 Mobile: (+55 155) 5514-087790 Mobile: (+55 155) 5541-351242 ___ -- 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] Queue member add
Queue members can be SIP channel names, including ones that are reachable at remote destination URIs, if that is what you re asking. e.g. member = SIP/[EMAIL PROTECTED] Queue members are made persistent in AstDB with the 'persistentmembers = yes' option and survive reboots. On Tue, 29 Jan 2008, Rob Schall wrote: Hopefully a fairly easy question for the group... I have a queue which should contain about 10 agents (it will be all the phones in the office). This office is remote, so I would like to add their sip phones into the queue remotely. Also, if the system ever gets reloaded or rebooted, I need those agents to remain in the queue. Question: 1) How do you remotely add agents to their respective queues and attach that to a specific sip phone? 2) How do you keep those phones in that queue even after the system reboots? Rob ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ -- 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's DANGEROUS Transfer CDR's
Disbaling transfers is an attractive option from my point of view but not from my customer's. Being able to transfer an incoming call from the receptionist to the required person is something businesses will consider changing provider for in my experience. The provider can disable transfers (which is what we do), but why can a PBX not still allow it? Our PBX customers all can do transferring... but that's because billing isn't needed THERE. The billing, if any, is done on our end, or their providers end.This really seems like a very small and moot point that is being blown up. If the receptionist needs to transfer the call, then she should be able to do that within the confines of her PBX... the transfer of her call should NEVER go back out her PBX back to the supplier, for if it does, her PBX now loses control of that call. ___ -- 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] ShoreTel - Asterisk Integration
Does anyone have experience using ShoreTel SIP trunks to integrate an Asterisk system? I am having trouble when the ShoreTel system transfers an incoming call from a SIP trunk to the voicemail system. From the SIP traffic, it looks like it negotiates a codec correctly, but once the RTP stream starts the call drops or there is no audio. I see errors in Asterisk such as: chan_sip.c:1944 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 104 (Critical Request) Has anyone run into this before or have any ideas? Thanks, Joe ___ -- 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 can I AIG?
Hey there, I've actually looked through the site a bunch and found some great information. The thing I'm missing, and I think it's because of my lack of experience with Asterisk and setting up a dial plan, is the multitude of ways/places where I can instantiate the AIG command. Do I have to control the entire call from beginning to end, or can I just call the Asterisk-Java class at certain points with certain parameters? Thanks! Evan From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Guilherme Loch Waltrick Góes Sent: Tuesday, January 29, 2008 1:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] When can I AIG? Have a look at asterisk-java.orghttp://asterisk-java.org. I has everything you need. On Jan 29, 2008 4:35 PM, Evan Ruff [EMAIL PROTECTED]mailto:[EMAIL PROTECTED] wrote: Hey Guys, I've been doing some research into the AGI-Java connector and was wondering if somebody could help me with my architecture. What I'd like to do, is kick off an external java class when a user: 1. Initiates an outgoing call 2. Hangs up the outgoing call 3. Has an incoming call 4. Hangs up the incoming call 5. Misses a call 6. Has a voicemail I'd also like to be able to access all the call details (user number, ext number, start time, end time) I see lot of references to the dial plan instantiating the AGI in the tutorials, but I was wondering if it is possible to use it in these scenarios? How would I need to connect these things up for each user? Sorry if this question is a big basic/elementary, but I'm just getting started with Asterisk and would really appreciate some help. Thanks a ton! Evan Ruff ___ -- 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 -- Guilherme Loch Góes Visite nossa loja virtual: http://www.shopvoip.com.br Notícias e Fórum sobre VoIP com software livre: http://www.asteriskexperts.com.br ___ -- 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's DANGEROUS Transfer CDR's
- Original Message From: C F [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, 29 January, 2008 8:05:00 PM Subject: Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's Grey, Just tested with 1.2.13 Asterisk always (blind or attd xfer) creates 2 records. A few points, NEVER rely on source as the billable number. Always use account codes. Match the lastdata field against dst fields to figure out that it was an xfer when doing the rating. The lastdata field will have the right number. That will work for blind transfers but not attended and even in the blind transfer case the CDR's still aren't correct you're relying on an informational field. Make the switch to the world's best email. Get the new Yahoo!7 Mail now. www.yahoo7.com.au/worldsbestemail ___ -- 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's DANGEROUS Transfer CDR's
Matt [EMAIL PROTECTED] writes: Asterisk is doing exactly as it should.. when it steps out of the media path, the CDR is also dropped, as asterisk is no longer responsible for that call. Even if Asterisk stays in the media path, the CDR's are dropped. It is an annoying problem. Hopefully the new CDR system provides a way to avoid it. It doesn't affect us so much because customers don't get to send transfer requests to our billing PBX's, but it's still silly. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] When does Asterisk REFER?
Philip Prindeville wrote: I'm trying to verify that my local PSTN's Coppercom switch operates correctly... and wanted to know how to get a call REFER'd to another end-point. I don't think Asterisk will ever generate a REFER, but the only possible way it could would be using the Transfer() application in the dialplan. -- 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's DANGEROUS Transfer CDR's
- Original Message From: Matt [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, 29 January, 2008 7:28:32 PM Subject: Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's Grey, I don't think you understand how transfers work. Let's take for example: USER-1 dials LOCATION A and then LOCATION B (referred to as 1,A,B). 1 Dials A and transfers the call to B. The call data is now NO LONGER in the asterisk path, therefore asterisk has nothing to do with the CDR. However, the call legs are still going out of the providers trunking. This is not a problem with asterisk, but a logic problem with you/providers dial-plan. Asterisk is doing exactly as it should.. when it steps out of the media path, the CDR is also dropped, as asterisk is no longer responsible for that call. Hi Matt, Sadly I understand all to well how transfers work. I've had to go over and over this for the last 12 months trying to find different ways of handling it. I'm talking about blind and attended call transfers here not IAX or any other kind. We are not taking Asterisk out of the media path and even if we were you wouldn't want to be losing CDR's from a provider's point of view, whoever set the call up is still paying for it regardless of where the media has been re-invited to. Out of the 8 Asterisk based providers I have tested 3 have this issue and the other 5 don't support transfers. It's dead simple for anyone to test. Find an Asterisk provider that supports transfers, connect with the xten, do a blind or attended transfer and check the CDR's. Call a free or cheap destination as the first leg of your transfer and the expensive destination second. You'll be pleasantly suprised at the bill! Regards, Greyman. Make the switch to the world's best email. Get the new Yahoo!7 Mail now. www.yahoo7.com.au/worldsbestemail ___ -- 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's DANGEROUS Transfer CDR's
Grey, Just tested with 1.2.13 Asterisk always (blind or attd xfer) creates 2 records. A few points, NEVER rely on source as the billable number. Always use account codes. Match the lastdata field against dst fields to figure out that it was an xfer when doing the rating. The lastdata field will have the right number. On Jan 29, 2008 2:28 PM, Matt [EMAIL PROTECTED] wrote: Grey, I don't think you understand how transfers work. Let's take for example: USER-1 dials LOCATION A and then LOCATION B (referred to as 1,A,B). 1 Dials A and transfers the call to B. The call data is now NO LONGER in the asterisk path, therefore asterisk has nothing to do with the CDR. However, the call legs are still going out of the providers trunking. This is not a problem with asterisk, but a logic problem with you/providers dial-plan. Asterisk is doing exactly as it should.. when it steps out of the media path, the CDR is also dropped, as asterisk is no longer responsible for that call. On Jan 29, 2008 12:48 PM, Grey Man [EMAIL PROTECTED] wrote: - Original Message From: Richard Revels [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, 29 January, 2008 12:21:16 PM Subject: Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's It's not Asterisk, it's SIP. Transfer takes the signaling off the Asterisk box. In features.conf, replace blind transfer with a call to a macro. Then redo your dialplan with the 'g' option on inward dial commands. When the called party uses the transfer command, your macro should read the digits to call and then store them in the db, a unique global, or GROUP () variable. Then it should hang up. This will cause the calling leg to exit the dial command to the next priority which should be a check of the variable. If digits are present, use the dial command to call them at your provider. No fuss, no muss. You should make sure the peer entry for the outbound side includes canreinvite=yes so only the signaling remains on your box and the media is invited off. You should also ignore calls to your macro that hit from the inbound call leg. Just return immediately and neither side will ever know the inbound call leg left for a moment. Sent from my iPhone Hi Richard, I'm not actually sure we're talking about the same thing here. It's not transfers I have a problem with it's the CDRs the transferred calls end up generating. In this case I am the provider and transfers through our Asterisk servers work fine it's just that we can't properly bill for them. Regards, Greyman. Make the switch to the world's best email. Get the new Yahoo!7 Mail now. www.yahoo7.com.au/worldsbestemail ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's
Grey, I don't think you understand how transfers work. Let's take for example: USER-1 dials LOCATION A and then LOCATION B (referred to as 1,A,B). 1 Dials A and transfers the call to B. The call data is now NO LONGER in the asterisk path, therefore asterisk has nothing to do with the CDR. However, the call legs are still going out of the providers trunking. This is not a problem with asterisk, but a logic problem with you/providers dial-plan. Asterisk is doing exactly as it should.. when it steps out of the media path, the CDR is also dropped, as asterisk is no longer responsible for that call. On Jan 29, 2008 12:48 PM, Grey Man [EMAIL PROTECTED] wrote: - Original Message From: Richard Revels [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, 29 January, 2008 12:21:16 PM Subject: Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's It's not Asterisk, it's SIP. Transfer takes the signaling off the Asterisk box. In features.conf, replace blind transfer with a call to a macro. Then redo your dialplan with the 'g' option on inward dial commands. When the called party uses the transfer command, your macro should read the digits to call and then store them in the db, a unique global, or GROUP () variable. Then it should hang up. This will cause the calling leg to exit the dial command to the next priority which should be a check of the variable. If digits are present, use the dial command to call them at your provider. No fuss, no muss. You should make sure the peer entry for the outbound side includes canreinvite=yes so only the signaling remains on your box and the media is invited off. You should also ignore calls to your macro that hit from the inbound call leg. Just return immediately and neither side will ever know the inbound call leg left for a moment. Sent from my iPhone Hi Richard, I'm not actually sure we're talking about the same thing here. It's not transfers I have a problem with it's the CDRs the transferred calls end up generating. In this case I am the provider and transfers through our Asterisk servers work fine it's just that we can't properly bill for them. Regards, Greyman. Make the switch to the world's best email. Get the new Yahoo!7 Mail now. www.yahoo7.com.au/worldsbestemail ___ -- 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] speex, ilbc and g729 codecs
Hi List; Anyone tried to use speex, ilbc and g729 and come back with a preferred one in the quality? Regards Bilal 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] speex, ilbc and g729 codecs
bilal ghayyad wrote: Hi List; Anyone tried to use speex, ilbc and g729 and come back with a preferred one in the quality? Regards Bilal 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 Depends on what you're using it for. We've had excellent luck with iLBC for voice quality and g729 as well. Speex is better than GSM, but not spectacularly. The problem with all of them of course, is transcoding. They have less of a bandwidth footprint for acceptable quality, but more of a processor footprint. N. ___ -- 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] Quad core is not a good idea! (was: Asterisk on Dell PowerEdge 2950)
Thanks for the response. I have not bought it yet and here are the specs I am considering. Any comments before I make the purchase. *PowerEdge 2950 III* Date 1/29/2008 12:40:22 PM Central Standard Time Catalog Number 4 Retail 04Catalog Number / Description Product Code SKU Id *PowerEdge 2950 III*: Dual Core Intel(R) Xeon(R) 5160, 4MB Cache, 3.0GHz, 1333MHz FSB 29530W [223-4926] 1*Additional Processor*: Dual Core Intel(R) Xeon(R) 5160, 4MB Cache, 3.00GHz, 1333MHZ FSB 2PW30 [311-6222] 2*Memory*: 4GB 667MHz (4x1GB), Dual Ranked DIMMs 4G4D6D [311-6154] 3*Operating System*: No Operating System NOOS [420-6320] 11*Backplane*: 1x6 Backplane for 3.5-inch Hard Drives 1X6353 [311-7936] 18*Primary Controller*: SAS 6/iR Integrated, x6 Backplane S6IX6 [341-5941] 9*Hard Drive Configuration*: Integrated SAS/SATA, SAS 6/iR Integrated, No RAID 6SS [341-5720] 27 *Primary Hard Drive*: 250GB 7.2K RPM Serial ATA 3Gbps 3.5-in HotPlug Hard Drive 250S2 [341-3037] 8*2nd Hard Drive*: 250GB 7.2K RPM Serial ATA 3Gbps 3.5-in HotPlug Hard Drive 250S2 [341-3037] 23*Chassis Configuration*: No Rack Rails Included NORAIL [310-7411] 28*Riser Card*: Riser with 3 PCIe Slots PCIE [320-4607] 7*Power Supply*: Non-Redundant Power Supply NRPS3 [310-9895] 36*Bezel*: Rack Bezel BEZEL [313-3920] 17*Network Adapter*: Dual Embedded Broadcom(R) NetXtreme II 5708 Gigabit Ethernet NIC OBNIC [430-1764] 13*TCP/IP Offload Engine Enablement*: Broadcom TCP/IP Offload Engine Not Enabled NTOEKEY [430-1765] 6* Documentation*: Electronic Documentation and OpenManage CD Kit EDOCS [310-7415] 21 *CD/DVD Drive*: 24X IDE CD-ROM 24XCD [313-3932] 16*Floppy Drive*: No Floppy Drive for x6 Backplane NFDX6 [341-3685] 10*Mouse*: Mechanical Two-Button Mouse, USB USBMW [310-8171] 12*Server Accessories*: USB to PS2 Adapter for KVM Connectivity USBPS2C [310-6690] 57 *Hardware Support Services*: 3Yr BASIC SUPPORT: 5x10 HW-Only, 5x10 NBD Onsite U3OS [960-8162][960-8192][970-4070][984-1399][984-1417] 29*Installation Services*: No Installation Assessment NOINSTL [900-9997] 32 javascript:self.print(); Print javascript:self.print(); On 1/29/08, Massimo Nuvoli [EMAIL PROTECTED] wrote: broadband Voice ha scritto: Does anyone have any compatibilty issues with Dell *PowerEdge^TM 2950 III 2-Socket, Quad-Core 2U*? I plan on using this with the Digium T1 cards. Thanks. Consider 2 socket dual core CPU with more mhz, Asterisk is more IO than computation. The quad core CPU is lower in MHZ and all the CPUs share the same IO. The architecture design of the XEON board is wrong for really IO intensive apps, consider stepping up to a AMD arch. Also the 2950 is a 2U server that use IRQ sharing (WHY?), wrong with boards like the T1 Digium.. Bye. ___ -- 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] Change Default Voicemail Message
The issue is simple. You make the voicemail box be the same as the room number, then you get: Playback(Welcome-2-nursing-home) Voicemail(xxx,s) Voicemail plays you've reached room, xxx, please leave a message then beeps and records message. On Jan 7, 2008 5:29 PM, Daniel Cole [EMAIL PROTECTED] wrote: Thank you for your reply Trevor. Is there an easy way to achieve this with a computer generated voice? We do not wish to manually record the messages if possible, in the interests of a consistent message across all voicemail boxes. What would be the easiest way to do this? Also, can you please give me some pointers on how to get the voicemail to play the separate message before the normal voicemail message? I'm guessing it would be done with a custom voicemail content, but im not sure how to write it correctly. Many Thanks, Daniel Cole -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Trevor G. Hammonds *Sent:* Monday, 7 January 2008 3:52 PM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Change Default Voicemail Message Daniel, You could have Alison record a prompt Welcome to (nursing home) and re-record the prompt The person at extension... to be The person in room Then have your dialplan play the Welcome To... message before sending the call to voice mail. Then callers will hear Welcome to (Nursing Home). The person in room 5 is unavailable. Please leave your message... and if the resident has a recorded personal greeting or name, it would replace the The person... portion with either the resident's recorded name or greeting. Sincerely, Trevor Hammonds *From:* Daniel Cole *Sent:* Sunday, January 06, 2008 6:15 PM Hello List, I have a client (a nursing home) that we are looking at installing a trixbox for. One of the features that they would really like is a customized, standard voicemail recording for each of the residents rooms. We are looking for something along the lines of a voicemail recording like this: Welcome to (nursing home). You have reached room 5. Please leave a message after the tone. What would be the easiest way to get this to work. I have had a look at a few options, but I cant seem to find what I am after. Any help would be much appreciated. Thank You, Daniel Cole ___ --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] When can I AIG?
Have a look at asterisk-java.org. I has everything you need. On Jan 29, 2008 4:35 PM, Evan Ruff [EMAIL PROTECTED] wrote: Hey Guys, I've been doing some research into the AGI-Java connector and was wondering if somebody could help me with my architecture. What I'd like to do, is kick off an external java class when a user: 1. Initiates an outgoing call 2. Hangs up the outgoing call 3. Has an incoming call 4. Hangs up the incoming call 5. Misses a call 6. Has a voicemail I'd also like to be able to access all the call details (user number, ext number, start time, end time) I see lot of references to the dial plan instantiating the AGI in the tutorials, but I was wondering if it is possible to use it in these scenarios? How would I need to connect these things up for each user? Sorry if this question is a big basic/elementary, but I'm just getting started with Asterisk and would really appreciate some help. Thanks a ton! Evan Ruff ___ -- 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 -- Guilherme Loch Góes Visite nossa loja virtual: http://www.shopvoip.com.br Notícias e Fórum sobre VoIP com software livre: http://www.asteriskexperts.com.br ___ -- 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 can I AIG?
Hey Guys, I've been doing some research into the AGI-Java connector and was wondering if somebody could help me with my architecture. What I'd like to do, is kick off an external java class when a user: 1. Initiates an outgoing call 2. Hangs up the outgoing call 3. Has an incoming call 4. Hangs up the incoming call 5. Misses a call 6. Has a voicemail I'd also like to be able to access all the call details (user number, ext number, start time, end time) I see lot of references to the dial plan instantiating the AGI in the tutorials, but I was wondering if it is possible to use it in these scenarios? How would I need to connect these things up for each user? Sorry if this question is a big basic/elementary, but I'm just getting started with Asterisk and would really appreciate some help. Thanks a ton! Evan Ruff ___ -- 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] Queue member add
Hopefully a fairly easy question for the group... I have a queue which should contain about 10 agents (it will be all the phones in the office). This office is remote, so I would like to add their sip phones into the queue remotely. Also, if the system ever gets reloaded or rebooted, I need those agents to remain in the queue. Question: 1) How do you remotely add agents to their respective queues and attach that to a specific sip phone? 2) How do you keep those phones in that queue even after the system reboots? Rob ___ -- 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's DANGEROUS Transfer CDR's
Hi Matt, Sadly I understand all to well how transfers work. I've had to go over and over this for the last 12 months trying to find different ways of handling it. I'm talking about blind and attended call transfers here not IAX or any other kind. We are not taking Asterisk out of the media path and even if we were you wouldn't want to be losing CDR's from a provider's point of view, whoever set the call up is still paying for it regardless of where the media has been re-invited to. Out of the 8 Asterisk based providers I have tested 3 have this issue and the other 5 don't support transfers. It's dead simple for anyone to test. Find an Asterisk provider that supports transfers, connect with the xten, do a blind or attended transfer and check the CDR's. Call a free or cheap destination as the first leg of your transfer and the expensive destination second. You'll be pleasantly suprised at the bill! Regards, Greyman. Grey... I'm not debating that this is how it works. We provide wholesale VoIP and retail VoIP. Transfers are disabled on both of those. That was one of the first things we did... all media and calls stay in our system. If the company doesn't have transfers disabled, that is their own fault, and their loss. I know exactly what you are referring to, and technically I'd say Asterisk is still correct, because the leg of the call that billing was happening on (the sip client) is no longer there. ___ -- 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 does Asterisk REFER?
- Original Message From: Philip Prindeville [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, 29 January, 2008 7:11:01 PM Subject: [asterisk-users] When does Asterisk REFER? I was wondering under what conditions Asterisk will hand off a call to another switch. I'm trying to verify that my local PSTN's Coppercom switch operates correctly... and wanted to know how to get a call REFER'd to another end-point. Thanks, -Philip Hi Philip, You can use the transfer command in your dial plan apart from that I don't know of any case when Asterisk would initiate a REFER. Generally Asterisk is the one handling the REFER's in reponse to user's transfer requests. Regards, Aaron Make the switch to the world's best email. Get the new Yahoo!7 Mail now. www.yahoo7.com.au/worldsbestemail ___ -- 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's DANGEROUS Transfer CDR's
Grey Man wrote: That will work for blind transfers but not attended and even in the blind transfer case the CDR's still aren't correct you're relying on an informational field. I think there is an important point being missed here; Asterisk did not originate the concept of CDRs, nor did it specify what they contain or how they are to be collected and generated. CDRs have existed for decades before Asterisk was created, and they are a fairly well understood concept throughout the telephony switching industry. They were designed for billing, and in many telephony networks are still used for billing. However, CDRs were created before the users of those services had the ability to transfer calls, make three-way calls, make conference calls, and do other magical things. As such, there is no way in a CDR to represent this activity in any *complete* manner. Doing so will require a redesign of the CDR system, which Steve Murphy has already begun for Asterisk 1.6. As far as I am aware, everyone who builds a complete billing system for Asterisk and expects it to be accurate and reliable uses other means in addition to CDRs for collecting the information, or they restrict their users to not performing actions that will break the billing process. -- 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's DANGEROUS Transfer CDR's
- Original Message From: Matt [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, 29 January, 2008 8:39:25 PM Subject: Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's Grey... I'm not debating that this is how it works. We provide wholesale VoIP and retail VoIP. Transfers are disabled on both of those. That was one of the first things we did... all media and calls stay in our system. If the company doesn't have transfers disabled, that is their own fault, and their loss. I know exactly what you are referring to, and technically I'd say Disbaling transfers is an attractive option from my point of view but not from my customer's. Being able to transfer an incoming call from the receptionist to the required person is something businesses will consider changing provider for in my experience. There is no way Asterisk is correct with regards the CDR's produced by a transfer. That's what I'm hoping to get people to agree on and think about a change for. Asterisk is still correct, because the leg of the call that billing was happening on (the sip client) is no longer there. Correct? There are still two other calls that were initiated by the user why should they be dismissed because the first call hungup? In any case they are not dismissed entirely they are just combined and recorded inaccurately. Regards, Greyman Make the switch to the world's best email. Get the new Yahoo!7 Mail now. www.yahoo7.com.au/worldsbestemail ___ -- 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] External Incomming Call Directed PickUP
Someone could please help me ?
Regards,
Fernando
Fernando Berretta wrote:
Dear Lacy,
We are using Standard FreePbx installation and we are trying to direct
pickup all the calls with **EXT NUMBER.
[app-pickup]
include = app-pickup-custom
exten = _**.,1,Noop(Attempt to Pickup ${EXTEN:2} by ${CALLERID(num)})
exten = _**.,n,Pickup(${EXTEN:2})
This is the FreePbx configuration for call pickup but doesn't work for
calls which comes from users which are in other context neither
incoming calls from-trunk
Any help will be appreciated
Regards,
Fernando
Lacy Moore wrote:
My magic orb is on the fritz. Can you give some more info? What
extension is ringing? What are you dialing to pick up? What does
your conf files look like?
I think I might know what the problem is, but I need a little more
info. Read core show application Pickup carefully, and then re-read
it 3 or 4 more times. It seems odd at first, but then you catch on.
You are picking up the calling channel, not the called extension.
On Jan 25, 2008 5:28 PM, Fernando Berretta [EMAIL PROTECTED] wrote:
Hi,
I'm having problems with Directed PickUn and Asterisk 1.4.
Directed call pickup **EXT works ok with internal calls which are in the
same CONTEXT but,, with calls in which are from other context or
incoming calls from IVR this function doesn't work as is pointed in
http://bugs.digium.com/view.php?id=11639
I'm using FreePbx 2.3,, and dont know how to solve or workaround this
problem
Could some one please help me.
Best Regards,
Fernando
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Re: [asterisk-users] Queue member add
Rob Schall wrote: Hopefully a fairly easy question for the group... I have a queue which should contain about 10 agents (it will be all the phones in the office). This office is remote, so I would like to add their sip phones into the queue remotely. Also, if the system ever gets reloaded or rebooted, I need those agents to remain in the queue. Question: 1) How do you remotely add agents to their respective queues and attach that to a specific sip phone? You can add queue members remotely from the manager by using the QueueAdd manager action. If you can open an Asterisk CLI remotely, then you can use the queue add member command (add queue member if you're using 1.2) from there as well. 2) How do you keep those phones in that queue even after the system reboots? set persistentmembers=yes in queues.conf Rob Mark Michelson ___ -- 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] Source Based Call Routing
Hi List, I have a scenario that I want to try out (we potential have a client who would need this), but I am as of yet unable to find much help with it. What we want to do is have an asterisk box with a large number of extensions (1000+). This asterisk box will have approximately 3 SIP trunks setup back to providers. What we want to do is to be able to define groups of extensions that use specific outbound trunks. Approximately a third of the extensions will one the first trunk, a third the second trunk, and the rest will use the last trunk. We also need control over assigning with trunks the given extensions will use. Any suggestions on how to get this to work would be very much appreciated. Many Thanks, Daniel ___ -- 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] Source Based Call Routing
I would broker the dial-out requests through FastAGI and put the logic that examines extensions and implements the load balancing / distribution in there. On Wed, 30 Jan 2008, Daniel Cole wrote: Hi List, I have a scenario that I want to try out (we potential have a client who would need this), but I am as of yet unable to find much help with it. What we want to do is have an asterisk box with a large number of extensions (1000+). This asterisk box will have approximately 3 SIP trunks setup back to providers. What we want to do is to be able to define groups of extensions that use specific outbound trunks. Approximately a third of the extensions will one the first trunk, a third the second trunk, and the rest will use the last trunk. We also need control over assigning with trunks the given extensions will use. Any suggestions on how to get this to work would be very much appreciated. Many Thanks, Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ -- 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's DANGEROUS Transfer CDR's
- Original Message From: Kevin P. Fleming [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, 29 January, 2008 9:34:23 PM Subject: Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's Grey Man wrote: That will work for blind transfers but not attended and even in the blind transfer case the CDR's still aren't correct you're relying on an informational field. I think there is an important point being missed here; Asterisk did not originate the concept of CDRs, nor did it specify what they contain or how they are to be collected and generated. CDRs have existed for decades before Asterisk was created, and they are a fairly well understood concept throughout the telephony switching industry. They were designed for billing, and in many telephony networks are still used for billing. Hi Kevin, Thanks for responding. I'd actually prefer to use some form of real-time call control for billing within Asterisk but that's another story. For the half a dozen or so integrations we have done with PSTN carriers the CDRs are integral to the whole process. Arguably the biggest step in the whole interconnect process is matching up the CDRs for agreement. However, CDRs were created before the users of those services had the ability to transfer calls, make three-way calls, make conference calls, and do other magical things. As such, there is no way in a CDR to represent this activity in any *complete* manner. I understand there are likely to always remain certain things that CDR's cannot cope with but I don't think transfers fall into that category. Would there be anything wrong with recording a CDR for each end of a bridge instead of one CDR per bridge? If one end of the bridge changes, as in the case of a transfer, you get one CDR. When the bridge hangs up you get two CDR's which in fact does make sense as a bridge is two calls/channels. I'd be more than happy to produce call flows for: transfers, 3 way call, whatever else; with the exact CDRs if that would help to clarify things. Doing so will require a redesign of the CDR system, which Steve Murphy has already begun for Asterisk 1.6. Yes and thanks must go to Steve for delving into this very unglamorous area it's certainly not up there with video conferencing. The worrying thing though is the CDR's for attended transfers in 1.4 are now worse than they were in 1.2. I've read through Steve's blog posting on the new design and I think there are still some problems with the CDR scenarios. Using overlapping CDRs to determine if a transfer was in progress is fragile (what happens if simultaneous calls are supported) and apart from that the new CDRs will still don't provide enough information to bill all the call legs involved in a transfer. As far as I am aware, everyone who builds a complete billing system for Asterisk and expects it to be accurate and reliable uses other means in addition to CDRs for collecting the information, or they restrict their users to not performing actions that will break the billing process. That's fair enough I guess but there are quite a few people using Asterisk that have been relying exclusively on its CDRs that weren't aware of the inaccuracies. Certainly the 3 providers I found in the last two days weren't (I've emailed them now). I don't think it would be insurmountable to improve the CDR design in Asterisk. Maybe it won't get to a stage where it's perfect but if a new design was produced it would pave the way for those of us that this is a big deal for to assist in the implementation. Regards, Greyman. Make the switch to the world's best email. Get the new Yahoo!7 Mail now. www.yahoo7.com.au/worldsbestemail ___ -- 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] Do Asterisk requires audio codec to be installed?
No, there is no need for any audio codec to be installed. In that case it would just be the words worst B2B SIP UA. By default Asterisk comes with quite a few codecs. On Jan 29, 2008 5:03 AM, [EMAIL PROTECTED] wrote: Hi, Can you please tell me whether Asterisk requires any audio or video codec to be installed separately or it supports itself? Thanking you, Preeta P Please do not print this email unless it is absolutely necessary. Spread environmental awareness. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.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] Asterisk's DANGEROUS Transfer CDR's
- Original Message From: Matt [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, 29 January, 2008 9:24:14 PM Subject: Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's The provider can disable transfers (which is what we do), but why can a PBX not still allow it? Our PBX customers all can do transferring... but that's because billing isn't needed THERE. The billing, if any, is done on our end, or their providers end. This really seems like a very small and moot point that is being blown up. Depends how much it could cost you I guess :). If you're not supporting transfers it's a moot point if you are it's a bit more interesting. If the receptionist needs to transfer the call, then she should be able to do that within the confines of her PBX... the transfer of her call should NEVER go back out her PBX back to the supplier, for if it does, her PBX now loses control of that call. Our customer base is residential and small business. They don't want to either pay for or support another a PBX thats what they've come to us for in the first place a lot of the time. Regards, Greyman. Make the switch to the world's best email. Get the new Yahoo!7 Mail now. www.yahoo7.com.au/worldsbestemail ___ -- 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] softmodems bank for ast.
I think its totally possible. Most of these winmodems are just sound cards with a hybrid. On Jan 29, 2008 11:51 AM, Pepe Aracil [EMAIL PROTECTED] wrote: Hi. We need a full featured modem bank 20+ to attend data calls. IAXmodem only supports fax protocols because spandsp only support fax protocols. The idea is to do a IAX wrapper like IAXmodem but with a full featured (but propietary) softmodem library like PCTEL or linuxant. I hate astribank+hardmodems solution. Some body think that is possible? Sorry for my horrible English. ;) Bye. ___ -- 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] Source Based Call Routing
- Original Message From: Daniel Cole [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, 29 January, 2008 10:31:55 PM Subject: [asterisk-users] Source Based Call Routing Hi List, I have a scenario that I want to try out (we potential have a client who would need this), but I am as of yet unable to find much help with it. What we want to do is have an asterisk box with a large number of extensions (1000+). This asterisk box will have approximately 3 SIP trunks setup back to providers. What we want to do is to be able to define groups of extensions that use specific outbound trunks. Approximately a third of the extensions will one the first trunk, a third the second trunk, and the rest will use the last trunk. We also need control over assigning with trunks the given extensions will use. Any suggestions on how to get this to work would be very much appreciated. Hi Daniel, 3 different contexts in your dial plan would work. Assign each block of accounts (rather than extensions) to the context with the routes that they should use. To change an account from using one trunk to another it would be as simple as changing its context. Regards, Greyman. Make the switch to the world's best email. Get the new Yahoo!7 Mail now. www.yahoo7.com.au/worldsbestemail ___ -- 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's DANGEROUS Transfer CDR's
On Jan 29, 2008 3:54 PM, Grey Man [EMAIL PROTECTED] wrote: - Original Message From: C F [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, 29 January, 2008 8:05:00 PM Subject: Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's Grey, Just tested with 1.2.13 Asterisk always (blind or attd xfer) creates 2 records. A few points, NEVER rely on source as the billable number. Always use account codes. Match the lastdata field against dst fields to figure out that it was an xfer when doing the rating. The lastdata field will have the right number. That will work for blind transfers but not attended and even in the blind transfer case the CDR's still aren't correct you're relying on an informational field. I tested it. It does so for both attended and blind. However, I only tested it with SIP xfers and not with Tt. Make the switch to the world's best email. Get the new Yahoo!7 Mail now. www.yahoo7.com.au/worldsbestemail ___ -- 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] chanspy does not pull the call back to asterisk after a reinvite
Franklin, Because ChanSpy() is a passive monitor, there is nothing about the implementation that would cause Asterisk to shunt the speech back to itself. Asterisk only does this in situations where it is out of the media path and needs to insinuate itself back into it for the purpose of generating media, such as on-hold music, IVR, etc. What you're wanting should, in my opinion, basically be submitted as a feature request. Perhaps the developers can add a flag to the ChanSpy() invocation repertoire to make this work. Cheers, -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ -- 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] chanspy does not pull the call back to asterisk after a reinvite
Hello all, I am allowing a reinvite between a snom 320 phone and a SIP gateway to take load off my Asterisk server. When I put the caller on hold, for example, Asterisk successfully reinserts itself into the rtp stream to play music on hold to the caller, but when I do a chanspy Asterisk does not seem to pull the call back. If I am spying on a channel when the call build up happens the reinvite never occurs and it works, but I cannot jump in and spy on a call in progress once the reinvite has happened. Has anyone run into this issue any maybe have a solution, or does anyone know of a good way to get that call back onto the Asterisk switch from another extension prior to calling chanspy? Thanks much, Franklin Webb -- Franklin Webb Asst Project Manager Inter Medi@ Marketing Solutions 610-701-9670 [EMAIL PROTECTED] ___ -- 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] Source Based Call Routing
You can also look at routing based on number ranges (if you keep the separate numbers in separate number ranges) but I would guess that this is not going to suit your needs. Maybe storing all the accounts in mysql (realtime) would also be a good planh. PaulH On Wed, 2008-01-30 at 09:31 +1100, Daniel Cole wrote: Hi List, I have a scenario that I want to try out (we potential have a client who would need this), but I am as of yet unable to find much help with it. What we want to do is have an asterisk box with a large number of extensions (1000+). This asterisk box will have approximately 3 SIP trunks setup back to providers. What we want to do is to be able to define groups of extensions that use specific outbound trunks. Approximately a third of the extensions will one the first trunk, a third the second trunk, and the rest will use the last trunk. We also need control over assigning with trunks the given extensions will use. Any suggestions on how to get this to work would be very much appreciated. Many Thanks, Daniel ___ -- 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] Source Based Call Routing
I would still say the easiest thing by far is to introduce a mediator in the dial plan that is far more intelligent and extensible than the dial plan logic itself. Enter FastAGI. Then you can just do it ... however you want. On Wed, 30 Jan 2008, Paul Hales wrote: You can also look at routing based on number ranges (if you keep the separate numbers in separate number ranges) but I would guess that this is not going to suit your needs. Maybe storing all the accounts in mysql (realtime) would also be a good planh. PaulH On Wed, 2008-01-30 at 09:31 +1100, Daniel Cole wrote: Hi List, I have a scenario that I want to try out (we potential have a client who would need this), but I am as of yet unable to find much help with it. What we want to do is have an asterisk box with a large number of extensions (1000+). This asterisk box will have approximately 3 SIP trunks setup back to providers. What we want to do is to be able to define groups of extensions that use specific outbound trunks. Approximately a third of the extensions will one the first trunk, a third the second trunk, and the rest will use the last trunk. We also need control over assigning with trunks the given extensions will use. Any suggestions on how to get this to work would be very much appreciated. Many Thanks, Daniel ___ -- 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ -- 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] Source Based Call Routing
Thank you Greg and Alex for your contribution. I will use your leads to see what I can get asterisk to do :) Many Thanks, Daniel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Grey Man Sent: Wednesday, 30 January 2008 9:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Source Based Call Routing - Original Message From: Daniel Cole [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, 29 January, 2008 10:31:55 PM Subject: [asterisk-users] Source Based Call Routing Hi List, I have a scenario that I want to try out (we potential have a client who would need this), but I am as of yet unable to find much help with it. What we want to do is have an asterisk box with a large number of extensions (1000+). This asterisk box will have approximately 3 SIP trunks setup back to providers. What we want to do is to be able to define groups of extensions that use specific outbound trunks. Approximately a third of the extensions will one the first trunk, a third the second trunk, and the rest will use the last trunk. We also need control over assigning with trunks the given extensions will use. Any suggestions on how to get this to work would be very much appreciated. Hi Daniel, 3 different contexts in your dial plan would work. Assign each block of accounts (rather than extensions) to the context with the routes that they should use. To change an account from using one trunk to another it would be as simple as changing its context. Regards, Greyman. Make the switch to the world's best email. Get the new Yahoo!7 Mail now. www.yahoo7.com.au/worldsbestemail ___ -- 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] Queue works with across server agent?
I am using Queue to handle some incoming calls. I wonder if the agent is across multiple servers, will this work? Thanks in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] When can I AIG?
Basically you initiate the call via a Originate method inn your Java App and bridge it to your AGI. Follow the documentation on the site and you should be in a good path. Best Regards, On Jan 29, 2008 7:01 PM, Evan Ruff [EMAIL PROTECTED] wrote: Hey there, I've actually looked through the site a bunch and found some great information. The thing I'm missing, and I think it's because of my lack of experience with Asterisk and setting up a dial plan, is the multitude of ways/places where I can instantiate the AIG command. Do I have to control the entire call from beginning to end, or can I just call the Asterisk-Java class at certain points with certain parameters? Thanks! Evan *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Guilherme Loch Waltrick Góes *Sent:* Tuesday, January 29, 2008 1:50 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] When can I AIG? Have a look at asterisk-java.org. I has everything you need. On Jan 29, 2008 4:35 PM, Evan Ruff [EMAIL PROTECTED] wrote: Hey Guys, I've been doing some research into the AGI-Java connector and was wondering if somebody could help me with my architecture. What I'd like to do, is kick off an external java class when a user: 1. Initiates an outgoing call 2. Hangs up the outgoing call 3. Has an incoming call 4. Hangs up the incoming call 5. Misses a call 6. Has a voicemail I'd also like to be able to access all the call details (user number, ext number, start time, end time) I see lot of references to the dial plan instantiating the AGI in the tutorials, but I was wondering if it is possible to use it in these scenarios? How would I need to connect these things up for each user? Sorry if this question is a big basic/elementary, but I'm just getting started with Asterisk and would really appreciate some help. Thanks a ton! Evan Ruff ___ -- 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 -- Guilherme Loch Góes Visite nossa loja virtual: http://www.shopvoip.com.br Notícias e Fórum sobre VoIP com software livre: http://www.asteriskexperts.com.br ___ -- 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 -- Guilherme Loch Góes Visite nossa loja virtual: http://www.shopvoip.com.br Notícias e Fórum sobre VoIP com software livre: http://www.asteriskexperts.com.br ___ -- 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] Queue - ${ANSWEREDTIME}
How to make ${ANSWEREDTIME} to work with Queue, so when the user hangs up, I
can calculate how much time the each call lasts?
Thanks
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Re: [asterisk-users] When can I AIG?
I just save a bunch of money on my car insurance by switching to AIG!!! Sorry, couldn't resist, Steve Totaro On Jan 29, 2008 8:11 PM, Guilherme Loch Waltrick Góes [EMAIL PROTECTED] wrote: Basically you initiate the call via a Originate method inn your Java App and bridge it to your AGI. Follow the documentation on the site and you should be in a good path. Best Regards, On Jan 29, 2008 7:01 PM, Evan Ruff [EMAIL PROTECTED] wrote: Hey there, I've actually looked through the site a bunch and found some great information. The thing I'm missing, and I think it's because of my lack of experience with Asterisk and setting up a dial plan, is the multitude of ways/places where I can instantiate the AIG command. Do I have to control the entire call from beginning to end, or can I just call the Asterisk-Java class at certain points with certain parameters? Thanks! Evan From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Guilherme Loch Waltrick Góes Sent: Tuesday, January 29, 2008 1:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] When can I AIG? Have a look at asterisk-java.org. I has everything you need. On Jan 29, 2008 4:35 PM, Evan Ruff [EMAIL PROTECTED] wrote: Hey Guys, I've been doing some research into the AGI-Java connector and was wondering if somebody could help me with my architecture. What I'd like to do, is kick off an external java class when a user: 1. Initiates an outgoing call 2. Hangs up the outgoing call 3. Has an incoming call 4. Hangs up the incoming call 5. Misses a call 6. Has a voicemail I'd also like to be able to access all the call details (user number, ext number, start time, end time) I see lot of references to the dial plan instantiating the AGI in the tutorials, but I was wondering if it is possible to use it in these scenarios? How would I need to connect these things up for each user? Sorry if this question is a big basic/elementary, but I'm just getting started with Asterisk and would really appreciate some help. Thanks a ton! Evan Ruff ___ -- 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 -- Guilherme Loch Góes Visite nossa loja virtual: http://www.shopvoip.com.br Notícias e Fórum sobre VoIP com software livre: http://www.asteriskexperts.com.br ___ -- 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 -- Guilherme Loch Góes Visite nossa loja virtual: http://www.shopvoip.com.br Notícias e Fórum sobre VoIP com software livre: http://www.asteriskexperts.com.br ___ -- 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] chanspy does not pull the call back to asterisk after a reinvite
On Jan 29, 2008 5:55 PM, Alex Balashov [EMAIL PROTECTED] wrote: Franklin, Because ChanSpy() is a passive monitor, there is nothing about the implementation that would cause Asterisk to shunt the speech back to itself. Asterisk only does this in situations where it is out of the media path and needs to insinuate itself back into it for the purpose of generating media, such as on-hold music, IVR, etc. What you're wanting should, in my opinion, basically be submitted as a feature request. Perhaps the developers can add a flag to the ChanSpy() invocation repertoire to make this work. Cheers, -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 Alex, he was not asking why, it is obvious he knows why. He was asking for a solution or idea on how to work around this issue. Are you using Sangoma cards? If so, I might have a very good answer for you, as well as another very possible different solution. Both would be outside of Asterisk so some kind of magic would have to happen to associate the call being spied on to the channel but that should not be that difficult if you even need it. Another solution is to track down the code referenced here http://bugs.digium.com/view.php?id=9888 and modify chanspy to do a reinvite back to asterisk before starting the spy. Anyways, I am sure it can be done. The question is how much time is it worth to make it happen. Maybe we should meet for lunch this week. I can meet you in cow country or Philly if you want, your choice. I have to go to both this week anyways and would like to catch up with things since Astricon. Thanks, Steve Totaro ___ -- 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 gateway
Hello everybody Anyone, to know a gateway that works with nextel simm cards? I'm looking for them, in internet, but I did'n look. Best regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Best practice security for internet access to Asterisk
Hi All For the scenario of a single asterisk server that needs to serve clients on the net, as well as local office clients, I would be very interested in people's views of the best method to handle security to prevent net based attacks while still allowing the client access. Some of the challenges I see are: - preventing brute force and bot type attacks - monitoring for unusual events and notifying and acting appropriately - limiting damage if someone does get in - avoiding a Denial or degradation of service on your asterisk platform - making it easy for staff to use Some of this can be done with - firewall control - but its hard to limit where your clients will come from, besides restricting ports - scripts monitoring logs, I saw a recipe for checking password failures then blocking that ip after x failures, I imagine this could get quite sophisticated - using separate restrictions for offnet users but this kind of makes it harder for the staff members. - using a proxy in front of asterisk for SIP, to limit the available extensions and minimise the scanning impact on the asterisk box. I am hoping this could detect and prevent illegitimate or poorly formed requests or unknown user agents. Staff should be using a standard set. - using iax softclients to shift the attack requirements - I don't know much about how well these work - running all clients over a vpn e.g open vpn, but this is not so good for wireless handsets or other devices that can't do a vpn I am interested in all views and recommendations Thanks very much Cheers Duncan ___ -- 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 gateway
Any GSM 900/1800 gateway will work with a Nextel (US) SIM card. However I assume you actually want to register on a local iDEN network and not be roaming internationally (Nextel does not have any GSM roaming partners in the US) That is not possible. On Jan 29, 2008 9:34 PM, Carlos Rojas [EMAIL PROTECTED] wrote: Hello everybody Anyone, to know a gateway that works with nextel simm cards? I'm looking for them, in internet, but I did'n look. Best regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] sipsock_read: BAD! BAD! BAD!
Does anyone know the cause of these BAD BAD BAD messages? I think I lost all my calls when it happened too. We have nagios running against IAX and nagios reports that IAX is down. It would seem that the entire application locks up when this happens and calls are dropped. Connected to Asterisk 1.2.14 currently running on flexo (pid = 26846) Verbosity is at least 3 flexo*CLI show channels Channel Location State Application(Data) SIP/teleglobe-09f887 (None) Down(None) 1 active channel 12 active calls Jan 30 03:28:13 WARNING[2671]: channel.c:781 channel_find_locked: Avoided deadlock for '0x9f6a6f0', 10 retries! Jan 30 03:28:14 ERROR[26983]: chan_sip.c:11451 sipsock_read: We could NOT get the channel lock for SIP/teleglobe-09f83250 - Call ID [EMAIL PROTECTED] Jan 30 03:28:14 ERROR[26983]: chan_sip.c:11452 sipsock_read: SIP MESSAGE JUST IGNORED: BYE Jan 30 03:28:14 ERROR[26983]: chan_sip.c:11453 sipsock_read: BAD! BAD! BAD! Jan 30 03:28:15 ERROR[26983]: chan_sip.c:11451 sipsock_read: We could NOT get the channel lock for SIP/teleglobe-09f83250 - Call ID [EMAIL PROTECTED] Jan 30 03:28:15 ERROR[26983]: chan_sip.c:11452 sipsock_read: SIP MESSAGE JUST IGNORED: BYE Jan 30 03:28:15 ERROR[26983]: chan_sip.c:11453 sipsock_read: BAD! BAD! BAD! Jan 30 03:28:16 ERROR[26983]: chan_sip.c:11451 sipsock_read: We could NOT get the channel lock for SIP/teleglobe-09f83250 - Call ID [EMAIL PROTECTED] Jan 30 03:28:16 ERROR[26983]: chan_sip.c:11452 sipsock_read: SIP MESSAGE JUST IGNORED: BYE Jan 30 03:28:16 ERROR[26983]: chan_sip.c:11453 sipsock_read: BAD! BAD! BAD! Jan 30 03:28:19 ERROR[26983]: chan_sip.c:11451 sipsock_read: We could NOT get the channel lock for SIP/teleglobe-09f83250 - Call ID [EMAIL PROTECTED] Jan 30 03:28:19 ERROR[26983]: chan_sip.c:11452 sipsock_read: SIP MESSAGE JUST IGNORED: BYE Doug. Looking for last minute shopping deals? Find them fast with Yahoo! Search. 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
Re: [asterisk-users] SIP DTMF Troubleshoot
Everything seems find on my end. Here's the setup:
Linksys SPA922 - Asterisk 1.4 --- Quintum T1 gateway
Between Asterisk and Quintum if I use G729 RFC2833 DTMF works with no
issues, however if I use uLaw this is where there is a problem. For
some reason the Quintum gateway does not support uLaw + RFC2833.
Also does not matter if I use Asterisk 1.2 or a grandstream or the
proverbial SIP tin can; The scenario is always the same.
On Jan 28, 2008 7:03 PM, Alex Balashov [EMAIL PROTECTED] wrote:
I think your best bet is to do a packet capture and look for RTP packets
with an RTP Event payload (rtpevent display filter).
On Mon, 28 Jan 2008, Andrew Joakimsen wrote:
How can I debug SIP DTMF? If I do 'sip set debug peer xyz' I see no
messages related to DTMF... or if I just do a global SIP debug for
that matter I am using RFC DTMF but it's not being passed to the
PSTN and I need to debug this further. I've tried to increase the
verbosity and the debug ('set debug n') and that didn't help either. I
assume this is because even RFC2833 sends the DTMF as RTP which
wouldn't show up anyways but how to troubleshoot DTMF issues?
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--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: +1-678-954-0670
Direct : +1-678-954-0671
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Re: [asterisk-users] Queue - ${ANSWEREDTIME}
It doesn't actually work at all - I tried, and even logged a bug with
digium with no luck. :(
Are the queue logs not quite good enough?
PaulH
On Tue, 2008-01-29 at 17:20 -0800, Johnny Tam wrote:
How to make ${ANSWEREDTIME} to work with Queue, so when the user hangs
up, I can calculate how much time the each call lasts?
Thanks
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Re: [asterisk-users] Source Based Call Routing
Daniel,
attach a dialplan variable to each extension using setvar
in sip.conf:
[6318]
type=friend
username=6318
secret=xx
host=dynamic
nat=no
dtmfmode=rfc2833
qualify=0
amaflags=billing
disallow=all
allow=alaw
allow=ulaw
canreinvite=no
context=phone
setvar=__usetrunk=1
you can use the ${usetrunk} variable in your dialpan.
Ron
Daniel Cole wrote:
Hi List,
I have a scenario that I want to try out (we potential have a client who would
need this), but I am as of yet unable to find much help with it.
What we want to do is have an asterisk box with a large number of extensions
(1000+). This asterisk box will have approximately 3 SIP trunks setup back to
providers. What we want to do is to be able to define groups of extensions that
use specific outbound trunks.
Approximately a third of the extensions will one the first trunk, a third the
second trunk, and the rest will use the last trunk. We also need control over
assigning with trunks the given extensions will use.
Any suggestions on how to get this to work would be very much appreciated.
Many Thanks,
Daniel
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smime.p7s
Description: S/MIME Cryptographic Signature
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[asterisk-users] Problem with DTMF dialing
Hi all I have a small problem here. I asked this question on another asterisk mailing list, but nobody seemed to be able to help me there. We are running * Asterisk 1.4.17 * Libpri 1.4.3 * Zaptel 1.4.8 on a 1.6 dual core, 2GB ram and a digium TDM800P wildcard, hardware echo cancelation and a quad FXO card. We have 4 analog lines, one of which is a Cellphone line for least cost routing. The problem I am having is dialing out using DTMF signalling. At the moment I am making do with Pulse dialing through the 3 analog lines. I can recieve calls on the Cellphone line without any problems, but cant dial out through it, as a cellphone cant do pulse dialing. I have run ztmonitor 1 -f gains, where 1 is the zap channel where the cellphone is located, while dialing the number 072 031 1294. I then went to audacity, on my own pc, and converted the raw file into mp3 format, which is available for download at http://www.iancoetzee.za.net/tone_dial.mp3. After listening to the playback I concluded that the DTMF signals being sent is totally wrong. The relevant pieces of my configs are below Your help in this matter will be greatly apreciated. Regards Ian -- www.vddi.co.za http://www.vddi.co.za/ I Coetzee IT Technician Telephone : 012 664 2300 Cellphone : 079 522 6519 Fax : 012 644 2902 E-mail : [EMAIL PROTECTED] Skype : vddb_igcoetzee */etc/asterisk/zapata.conf* ; Span 1: WCTDM/0 Wildcard TDM800P Board 1 (MASTER) ;;; line=1 WCTDM/0/0 ;Cellphone signalling=fxs_ks callerid=asreceived context=incoming_calls callerid= group=2 busydetect=yes usecallerid=yes hidecallerid=no callwaiting=no threewaycalling=yes transfer=yes echocancel=yes echotraining=yes pulsedial=no callprogress=yes busycount=5 toneduration=500 subscribecontext=GXP_BLF overlapdial=no channel = 1 ;;; line=2 WCTDM/0/1 ;Landline signalling=fxs_ks callerid=asreceived context=incoming_calls callerid= group=1,2 busydetect=yes usecallerid=yes hidecallerid=no callwaiting=no threewaycalling=yes transfer=yes echocancel=yes echotraining=yes pulsedial=yes callprogress=yes busycount=5 toneduration=300 subscribecontext=GXP_BLF channel = 2 */etc/zaptel.conf* # Autogenerated by /usr/sbin/zapconf on Wed Jan 16 12:23:09 2008 -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # Span 1: WCTDM/0 Wildcard TDM800P Board 1 (MASTER) fxsks=1 fxsks=2 fxsks=3 fxsks=4 # channel 5, WCTDM/0/4, no module. # channel 6, WCTDM/0/5, no module. # channel 7, WCTDM/0/6, no module. # channel 8, WCTDM/0/7, no module. # Global data loadzone= za defaultzone = za* * ___ -- 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
