[asterisk-users] Performance Issues Degradation After 6 Calls
I am using Asterisk and A2billing Calling Card Platform and after the 6th call the quality starts to degrade. The way it set up is the user calls into the system then dial out so I have 12 channels being used up but 6 active calls. Here are my specs Asterisk SVN-branch-1.4-r79142 on a i686 running Linux Fedora 6, Pentium 4 Hyper-Threading, 64 bit, 1GB of RAM, 80 GB Sata Drive, bandwidth 4 Mbps (1300GB/Throughput) burstable to 100Mbps. I am planning on upgrading to Intel Core 2 Duo with a clock speed of 1.8GHZand 2GB Ram. Does anyone have similar situation or advice? 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] SIP Channel jitter buffer issue
Hi,
Forgot to mention that I am using Asterisk version 1.4.15 running on RHEL
3.0 server.
Regards,
Mayur
_
From: Mayur [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 26, 2007 7:03 PM
To: 'asterisk-users@lists.digium.com'
Subject: SIP Channel jitter buffer issue
Hi,
I have a SIP client which is registered to asterisk. Asterisk is
registered to a SIP trunk and also handles the media. Now since my client
has some issues in its RTP Tx, which seems to have some amount of jitter
(mean jitter as per ethereal trace is about 17ms, max jitter is 20 ms and
max delta is 85 ms), to over come that I have enabled jitter buffer in the
SIP channel by setting sip.conf parameters jenable=yes, jbforce=yes,
jbmaxsize=200 and jbimpl=fixed. However on setting these parameters I am
unable to hear on the trunk side. From the jitter logs as given below, I can
see audio frames being dropped:
JB_PUT {now=1130}: Dropped frame with ts=21125 and len=20
JB_GET {now=1130}: now next=2121
JB_GET {now=1142}: now next=2121
JB_GET {now=1163}: now next=2121
JB_PUT {now=1181}: Dropped frame with ts=21132 and len=20
JB_GET {now=1181}: now next=2121
JB_GET {now=1183}: now next=2121
JB_PUT {now=1185}: Dropped frame with ts=21132 and len=20
JB_GET {now=1185}: now next=2121
I have tried increasing the jitter buffer from 200 to 1000 ms but with same
result.
Am I missing anything here? How can I determine what is causing asterisk to
drop the audio frames?
Regards,
Mayur
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[asterisk-users] application not load
hi, all I creat new application app_myapp.c for asterisk 1.4.15. I add this in asterisk/apps dir. to load. after compiling asterisk app_myapp.o and app_myapp.so has been created but when i run show applications at cli . my application not displayed. what's wrong??? any suggestion!!! thanks Bhrugu Mehta ___ --Bandwidth and Colocation Provided 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] Grandtream Conference issue
Hi, I'm using Grandstream IP phone GXP2000, with Asterisk 1.4.15 I'm using g729 codec and want to use only this codec for the calls. My normal calls are going fine. But issue is coming when I'm using the conference from the Line1 and Line2 Option. When I'm initiating the conference at that time, IP phone is sending the G711ulaw for the conference call, while in my phone I've set the all codec option to PCMU only. Due to this I'm facing issue. Any solution for this problem, please let me know. Regards, Keshav Regards, Kesh Lets change the future...lets change the world. - Never miss a thing. Make Yahoo your homepage.___ --Bandwidth and Colocation Provided 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] zap transfer
hi, all I want to transfer my zap incoming call to another hard phone. is there any way to transfer call. our company is using CORAL EPBX. thnks for any suggestion Bhrugu Mehta ___ --Bandwidth and Colocation Provided 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] Samsung iDCS 500R2 PRI Asterisk 1.4.*
william, post your dialplan section for outbound, sip.conf minus passwords, and CLI output so we can see what is going on... make sure you set debugging higher or set sip debugging on for iDCS, assuming iDCS is a sip provider, of course daveC William Stillwell (Ki4swy) wrote: I'm having trouble getting an Asterisk server to make outbound calls on my iDCS Trunks. I have idcs station to asterisk station working I have asterisk station to idcs station working However, I am unable to get Asterisk to utilize any outbound trunks on my iDCS Anybody have any ideas? Sent via the WebMail system at kotbh.net ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- My wife's sister is in California. I should buy her a Videophone2008! Truly, The Next Best Thing to Being There! -- WorldWideVideoPhones.com 856.380.0894 ___ --Bandwidth and Colocation Provided 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] Samsung iDCS 500R2 PRI Asterisk 1.4.*
I'm having trouble getting an Asterisk server to make outbound calls on my iDCS Trunks. I have idcs station to asterisk station working I have asterisk station to idcs station working However, I am unable to get Asterisk to utilize any outbound trunks on my iDCS Anybody have any ideas? Sent via the WebMail system at kotbh.net ___ --Bandwidth and Colocation Provided 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] Marry Christmas and Happy New Year!!!
Happy New Year to Team !! John Nguyen, Kerry S [EMAIL PROTECTED] wrote: Thank you. We new what you meant. And a Merry Christmas and a Happy New Year to you too. On Dec 26, 2007 7:33 PM, Josu� Conti [EMAIL PROTECTED] wrote: Yep, excuse me I typed quickly. But the ideia was this to desire to a Merry Christmas and Happy New Year. Best Regards Josu� 2007/12/26, Doug Lytle [EMAIL PROTECTED]: Anthony Francis wrote: Not to be a nitpicker (well actually that is precisely what I am doing) but isn't it Merry? That it is, but for those that aren't native English speakers, I look the other way most of the time. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided 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 - Never miss a thing. Make Yahoo your homepage.___ --Bandwidth and Colocation Provided 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] ISDN BRI support with HFC-PCI cards?
On Thu, Dec 27, 2007 at 04:00:49PM +0100, Jaap Winius wrote: Quoting Tzafrir Cohen [EMAIL PROTECTED]: cat /proc/zaptel/* Span 1: ZTHFC1 HFC-S PCI A ISDN card 0 [TE] layer 1 DEACTIVATED (F4) AMI/CCS 1 ZTHFC1/0/1 Clear (In use) 2 ZTHFC1/0/2 Clear (In use) 3 ZTHFC1/0/3 HDLCFCS (In use) Span 2: ZTHFC2 HFC-S PCI A ISDN card 1 [TE] layer 1 ACTIVATED (F7) AMI/CCS 4 ZTHFC2/0/1 Clear (In use) 5 ZTHFC2/0/2 Clear (In use) 6 ZTHFC2/0/3 HDLCFCS (In use) Looks like it's already configured and used by Asterisk. Indeed. It would appear that Asterisk now recognizes these cards. Of course, I now have another set of problems, but I'll ask about those in a new thread. However, I believe that the ports are disconnected, right? Physically? One had a cable in it, the other one didn't. Sorry, and it is clearly stated above. I have just misread it. So basically dial to Zap/g0/NUMBER and it should dial it to and it should dial it to your provider. Using pridialplan=unknown in zapata.conf may also help. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN BRI support with HFC-PCI cards?
On Thu, Dec 27, 2007 at 06:03:09PM +1300, Matt Riddell wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Can Zaptel support HFC cards now (i.e. without mISDN?) Zaptel drivers for PCI HFC-S cards (cards that use the single-port chips of Cologne Chips) is zaphfc from BRIstuff, or vzaphfc. See http://bristuff.org/ . You'll also need support for BRI, PtMP and such in libpri and Asterisk. That generally requires the libpri and Asterisk patches from bristuff. If so, does that include the Digium b410p card? That card is uses the Cologne Chips HFC-4S chip. qozap from BRIstuff is generally a driver for a similar hardware, but I assume it would need some adaptations. For instance, patches to adapt that driver to support the very similar BeroNet cards can be found at http://blog.eth0.cc/zaptel-patchwork . I have no idea what changes would be required to adapt the b410p card to use qozap. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandtream Conference issue
keshaw, did you set your sip.conf to only allow g729? disallow=all allow=g729 I don't use g729 so the allow= may not be the correct syntax... here is the config I uise: disallow=all allow=ulaw allow=gsm allow=alaw daveC Keshav K. wrote: Hi, I'm using Grandstream IP phone GXP2000, with Asterisk 1.4.15 I'm using g729 codec and want to use only this codec for the calls. My normal calls are going fine. But issue is coming when I'm using the conference from the Line1 and Line2 Option. When I'm initiating the conference at that time, IP phone is sending the G711ulaw for the conference call, while in my phone I've set the all codec option to PCMU only. Due to this I'm facing issue. Any solution for this problem, please let me know. Regards, Keshav Regards, Kesh Lets change the future...lets change the world. Never miss a thing. Make Yahoo your homepage. http://us.rd.yahoo.com/evt=51438/*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 No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.17.7/1194 - Release Date: 12/23/2007 05:27 PM -- My wife's sister is in California. I should buy her a Videophone2008! Truly, The Next Best Thing to Being There! -- WorldWideVideoPhones.com 856.380.0894 ___ --Bandwidth and Colocation Provided 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] ISDN BRI support with HFC-PCI cards?
Quoting Tzafrir Cohen [EMAIL PROTECTED]: cat /proc/zaptel/* Span 1: ZTHFC1 HFC-S PCI A ISDN card 0 [TE] layer 1 DEACTIVATED (F4) AMI/CCS 1 ZTHFC1/0/1 Clear (In use) 2 ZTHFC1/0/2 Clear (In use) 3 ZTHFC1/0/3 HDLCFCS (In use) Span 2: ZTHFC2 HFC-S PCI A ISDN card 1 [TE] layer 1 ACTIVATED (F7) AMI/CCS 4 ZTHFC2/0/1 Clear (In use) 5 ZTHFC2/0/2 Clear (In use) 6 ZTHFC2/0/3 HDLCFCS (In use) Looks like it's already configured and used by Asterisk. Indeed. It would appear that Asterisk now recognizes these cards. Of course, I now have another set of problems, but I'll ask about those in a new thread. However, I believe that the ports are disconnected, right? Physically? One had a cable in it, the other one didn't. Cheers, Jaap ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandtream Conference issue
On Thu, 2007-12-27 at 01:32 -0800, Keshav K. wrote: When I'm initiating the conference at that time, IP phone is sending the G711ulaw for the conference call, while in my phone I've set the all codec option to PCMU only. PCMU is another way of saying G711ulaw... they're the same codec. It's your basic 64kbps pulse-code modulated ulaw companded audio codec. (For more information on PCM audio and how it works, see the Digital Telephony section of Asterisk: The Future of Telephony, downloadable at http://www.asteriskdocs.org/ for free.) --- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR
Hi Steve, .. I'll try to sort all this out, and then I'll attack this problem. Hopefully, I get it all into svn before the next release of 1.4...! Just wondering if any new CDR functionality made it into the 1.4.16.2 release? I have looked through the ChangeLog for the 1.4.15 and 1.4.16.2 releases but didn't spot anything to do with changes in CDR handling. I for one still have big problems with CDR's for blind transfer and attended transfer calls. - For Blind Transfers the destination in the CDR ends up being the destination transferred to. This means you could ring a mobile and when finished blind transfer to a free number and get a free call. - For Attended Transfers it's all very confused. I get two CDR's both with the destination of the latest call in the transfer. I've got some users that have cottoned onto this and not only ring the low value destination second but then email in complaining about duplicate billing to try and get one of the CDR's refunded. On a separate note does anyone know how to block transfers on a SIP channel? I can block REFER requests from my SIP Proxy but I have to support some transfers so that's not an option. Regards, Greyman. - Original Message From: Steve Murphy [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, 15 October, 2007 6:22:45 PM Subject: Re: [asterisk-users] CDR Sorry! I've gotten some complaints on this; I will try this week to mod 1.4 so that you can choose to see the single-channel unanswered CDR's, in a new config file option. I've gotten complaints both ways, tho, so pardon me if I get a little confused about what users out there want from CDR's. My biggest trouble is that by forcing all channels to have a CDR at creation time solves problems with missing CDR's, but creates a problem by generating extra unanswered CDR's that weren't generated before... for instance, when you ring three phones via a dial command, you then get 3 CDR's, including the two phones that were rung, but not answered. Another problem is with Zap-based phones; you take the handset off-hook, and a channel is created and dialtone generated. If you hang up, you get a CDR there, also. I have not found an easy way to detect and drop these kinds of CDR's, as most folks really do not find them very useful. And, I've gotten a complaint that you end up with 'duplicate' CDR's, which is also an artifact of forcing all channels to have a CDR associated. If anybody thinks they have a magic spell that will calm down the CDR's, I will not mind the information at all! I worked all last week to try to iron out the 1.4 zap-transfer CDR issues. I have 12 cases I test with involving hook-flash and #-blindxfers, and so far, I've got 9 of the 12 working OK (as far as I'm concerned.), but I have 3 cases that come up with problems. For instance, if you hookflash, and dial a number, the CDR's will be different, if you hang up before the dialed party answers, versus hanging up afterwards... The diff between a blind xfer and an attended xfer (without the 3 way), I guess, but I lose the calling channel name... I'll try to sort all this out, and then I'll attack this problem. Hopefully, I get it all into svn before the next release of 1.4...! As far as xfers in 1.4 go, I'm trying to make sure that the source and destination channel names reflect the true dialing party, as this makes more sense from a billing perspective, at least to me. So, if A calls B, and B forwards the call to C, then the CDR's need to reflect a call from A to B, and a call from B to C, which you may or may not be seeing right now. AFAICT, transfers pretty much result in confused CDR's. I gave up totally on generating separate CDR's for any 3-ways that might occur. Such 3-ways will end up being billed to the dialing parties. Here's an interesting situation: A calls B, A then hookflashes, and then A calls C, and hookflashes again. It's now a 3-way call, between A, B, and C. A then drops out and B and C converse. My goal with this situation was to have two CDR's, one for A-B and one for A-c. Since A placed both calls, it seems only just that A pay for B's and C's conversation. Especially if A is in the US, for example, and B is in Uganda, and C is in Bangladesh! Getting the right info in the right field from the driver level is pretty tricky, and you can add the fact that there's definite timing issues at play. If I make changes to CDR's in channels A and B, near to the time a hangup occurs (and it's very commonly the case), I can end up with some pretty strange stuff happening! I found that adding debug logging statements to the driver can affect the way the CDR's are generated! I solved some of this by locking channels (which to me is pretty ugly, considering the number of locks involved). So, please, cut me some slack... and keep me informed about your happiness levels. I want this stuff to be good, solid,
Re: [asterisk-users] application not load
bhrugu, did you try and load it manually? Modules are compiled in to shared object (.so) files. They are installed to /usr/lib/asterisk/modules and can be turned on and off from loading by editing /etc/asterisk/modules.conf. Modules must include asterisk/modules.h. Modules must also export several functions. The following functions generally return 0 on success and non-zero on failure. Do not define any of these functions as static. http://www.lobstertech.com/doc/ast-12-func/#funcmod daveC Bhrugu Mehta wrote: hi, all I creat new application app_myapp.c for asterisk 1.4.15. I add this in asterisk/apps dir. to load. after compiling asterisk app_myapp.o and app_myapp.so has been created but when i run show applications at cli . my application not displayed. what's wrong??? any suggestion!!! thanks Bhrugu Mehta ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- My wife's sister is in California. I should buy her a Videophone2008! Truly, The Next Best Thing to Being There! -- WorldWideVideoPhones.com 856.380.0894 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How does Asterisk scale to 500-1000 phones?
Anyone have opinions on how well Asterisk scales to 500-1000 stations, in regards to reliability, system performance, as well as ease of management? Assume relatively low call volume; let's say two public network PRIs (47 DS0s). -- # Jesse Molina # The Translational Genomics Research Institute # http://www.tgen.org # Mail = [EMAIL PROTECTED] # Desk = 1.602.343.8459 # Cell = 1.602.323.7608 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No SMDI interfaces are available
Hi, I'm a brand newbie to asterisk trying to set it up for the first time and I can't get a softphone to connect, the connection times out. I had a trixbox pro install working, but I need more control and would like to learn to do it with asterisk. In /var/log/asterisk/messages I see: WARNING[17401] res_smdi.c: No SMDI interfaces are available to listen on, not starting SMDI listener. on startup. Can someone clue me in as to what this means? Is it the cause of my timeouts? Asterisk version 1.4.16.2, no hardware, zaptel is running with the ztdummy driver. thanks, --charlie -- Charles Farinella Appropriate Solutions, Inc. (www.AppropriateSolutions.com) [EMAIL PROTECTED] voice: 603.924.6079 fax: 603.924.8668 ___ --Bandwidth and Colocation Provided 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] No SMDI interfaces are available
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Charlie Farinella Sent: Thursday, December 27, 2007 11:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] No SMDI interfaces are available Hi, I'm a brand newbie to asterisk trying to set it up for the first time and I can't get a softphone to connect, the connection times out. I had a trixbox pro install working, but I need more control and would like to learn to do it with asterisk. In /var/log/asterisk/messages I see: WARNING[17401] res_smdi.c: No SMDI interfaces are available to listen on, not starting SMDI listener. on startup. Can someone clue me in as to what this means? Is it the cause of my timeouts? Asterisk version 1.4.16.2, no hardware, zaptel is running with the ztdummy driver. thanks, --charlie -- Charles Farinella Appropriate Solutions, Inc. (www.AppropriateSolutions.com) [EMAIL PROTECTED] voice: 603.924.6079 fax: 603.924.8668 Simplified Message Desk Interface (SMDI) defines a way for a phone system to provide voice-messaging systems with the information that the system needs to intelligently process incoming calls. Each time that the phone system routes a call, it sends an SMDI message through an EIA/TIA-232 connection to the voice-messaging system that tells it the line that it is using, the type of call that it is forwarding, and information about the source and destination of the call. So this is NOT the problem in your case. I would start by checking the basic TCP/ip connectivity b/w the machine running the * PBX and your Softphone. Try this: iptables -L are they any rules listed, by default many linux distros setup a basic fierewall for you that by default rejects or drops everything but ssh and/or http. Ping your PBX from your softphone PC. Is it reachable? Is asterisk running? (asterisk -rv) ___ --Bandwidth and Colocation Provided 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] No SMDI interfaces are available
On Thursday 27 December 2007, Alexander Lopez wrote: I would start by checking the basic TCP/ip connectivity b/w the machine running the * PBX and your Softphone. Try this: iptables -L are they any rules listed, by default many linux distros setup a basic fierewall for you that by default rejects or drops everything but ssh and/or http. Ping your PBX from your softphone PC. Is it reachable? Is asterisk running? (asterisk -rv) Duh!! Everything was set correctly except for the fact that eth0 was *not* set as a trusted device! Thanks for the pointer. --charlie -- Charles Farinella Appropriate Solutions, Inc. (www.AppropriateSolutions.com) [EMAIL PROTECTED] voice: 603.924.6079 fax: 603.924.8668 ___ --Bandwidth and Colocation Provided 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] CDR
On Thu, 2007-12-27 at 07:07 -0800, Grey Man wrote: Hi Steve, .. I'll try to sort all this out, and then I'll attack this problem. Hopefully, I get it all into svn before the next release of 1.4...! Just wondering if any new CDR functionality made it into the 1.4.16.2 release? I have looked through the ChangeLog for the 1.4.15 and 1.4.16.2 releases but didn't spot anything to do with changes in CDR handling. I for one still have big problems with CDR's for blind transfer and attended transfer calls. - For Blind Transfers the destination in the CDR ends up being the destination transferred to. This means you could ring a mobile and when finished blind transfer to a free number and get a free call. - For Attended Transfers it's all very confused. I get two CDR's both with the destination of the latest call in the transfer. I've got some users that have cottoned onto this and not only ring the low value destination second but then email in complaining about duplicate billing to try and get one of the CDR's refunded. On a separate note does anyone know how to block transfers on a SIP channel? I can block REFER requests from my SIP Proxy but I have to support some transfers so that's not an option. Regards, Greyman-- No real new functionality in 1.4, except a cdr.conf option that lets you control whether you see one-channel cdrs. I haven't been working on CDR's the last few months in favor of other projects that seem a little more urgent. Plus, I have some folks urging me NOT to proceed until some architectural issues are discussed, which might be wise. I have been working on one bug where I did make some substantive changes to how the CDR's are generated, but it is almost certain that these changes will only show up in trunk. I've reached the limit of what I can do in 1.4; it is simply impossible to do anything with CDR's in 1.4 without tearing the very fabric of time and space, and just plain getting everybody upset... at least, those who were not erased from existence by the tear... on a more serious note, the changes are intrusive enough, the behavior changes big enough, that they really don't qualify to be applied to a current release. It's a huge job! My past work was just in the ZAP channel driver code, and because it's so asynch, and all split up into different code, it's really tough to get the right pieces in the right places at the right time in the right way. What this all says is that I'm most likely NOT doing it the right way. And what worries me most is that there might not be any right way. But I'm still new to this, and will get back around to it hopefully fairly soon. murf Greyman. - Original Message From: Steve Murphy [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, 15 October, 2007 6:22:45 PM Subject: Re: [asterisk-users] CDR Sorry! I've gotten some complaints on this; I will try this week to mod 1.4 so that you can choose to see the single-channel unanswered CDR's, in a new config file option. I've gotten complaints both ways, tho, so pardon me if I get a little confused about what users out there want from CDR's. My biggest trouble is that by forcing all channels to have a CDR at creation time solves problems with missing CDR's, but creates a problem by generating extra unanswered CDR's that weren't generated before... for instance, when you ring three phones via a dial command, you then get 3 CDR's, including the two phones that were rung, but not answered. Another problem is with Zap-based phones; you take the handset off-hook, and a channel is created and dialtone generated. If you hang up, you get a CDR there, also. I have not found an easy way to detect and drop these kinds of CDR's, as most folks really do not find them very useful. And, I've gotten a complaint that you end up with 'duplicate' CDR's, which is also an artifact of forcing all channels to have a CDR associated. If anybody thinks they have a magic spell that will calm down the CDR's, I will not mind the information at all! I worked all last week to try to iron out the 1.4 zap-transfer CDR issues. I have 12 cases I test with involving hook-flash and #-blindxfers, and so far, I've got 9 of the 12 working OK (as far as I'm concerned.), but I have 3 cases that come up with problems. For instance, if you hookflash, and dial a number, the CDR's will be different, if you hang up before the dialed party answers, versus hanging up afterwards... The diff between a blind xfer and an attended xfer (without the 3 way), I guess, but I lose the calling channel name... I'll try to sort all this out, and then I'll attack this problem. Hopefully, I get it all into svn before the next release of 1.4...! As far as xfers in 1.4 go, I'm trying to make sure that the source and destination channel names reflect the true
Re: [asterisk-users] Grandtream Conference issue
Hi, I'm using Grandstream IP phone GXP2000, with Asterisk 1.4.15 I'm using g729 codec and want to use only this codec for the calls. My normal calls are going fine. But issue is coming when I'm using the conference from the Line1 and Line2 Option. When I'm initiating the conference at that time, IP phone is sending the G711ulaw for the conference call, while in my phone I've set the all codec option to PCMU only. Due to this I'm facing issue. Any solution for this problem, please let me know. Regards, Keshav Spoke with Grandstream about this awhile ago, they only support one G729 call per phone at this point. Not sure if the latest ROM works for this or not, I have yet to try it, but my guess is no, because it is not listed on their fixes. ___ --Bandwidth and Colocation Provided 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] VoIP 2008 : wish list and predictions
Don't miss the call this week, Friday at 12 Noon EST, 9 AM PST, 6PM Western Europe. About the conference (formerly Asterisk Users Conference, Asterisk is a registered trademark of Digium) * http://www.voipusersconference.org * There is a Flash player on the above page. If you hate Flash, you can download any recordings in mp3 format here: * http://food4wine.ning.com/conference * Two ways to call: * PSTN in the US, Call (724) 444-7444 * SIP sip:[EMAIL PROTECTED] After the call connects, enter the show id: 22622# and your_PIN# If you do not have a PIN, you can use 1#. If your PIN is your phone number, CID will be recognized. IRC #voip-users-conference on freenode.net Community blogs etc: http://food4wine.ning.com Have a happy, healthy and prosperous 2008 Randy ___ --Bandwidth and Colocation Provided 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] New voicemail app (supports many interfaces, including Audix)
We just completed a new implementation of voicemail for Asterisk. It's much cleaner than Comedian mail and can emulate several voicemail user interfaces, including Audix. It's a great replacement for Audix. All of the sounds/prompts are presently being re-recorded by a professional female voice. If you are interest in the app, let us know at [EMAIL PROTECTED] Justin Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ --Bandwidth and Colocation Provided 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] New voicemail app (supports many interfaces, including Audix)
Is it free and/or open source? Does it have a webpage? Thanks! Moj Justin Newman wrote: We just completed a new implementation of voicemail for Asterisk. It's much cleaner than Comedian mail and can emulate several voicemail user interfaces, including Audix. It's a great replacement for Audix. All of the sounds/prompts are presently being re-recorded by a professional female voice. If you are interest in the app, let us know at [EMAIL PROTECTED] Justin Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://us.rd.yahoo.com/evt=51734/*http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New voicemail app (supports many interfaces, including Audix)
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Justin Newman wrote: We just completed a new implementation of voicemail for Asterisk. It's much cleaner than Comedian mail and can emulate several voicemail user interfaces, including Audix. It's a great replacement for Audix. All of the sounds/prompts are presently being re-recorded by a professional female voice. If you are interest in the app, let us know at [EMAIL PROTECTED] I'm assuming that since you sent it to Asterisk Users (Non-Commercial Discussion) it is free. Is it also Open Source? What licence? - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHdDBvDQNt8rg0Kp4RArv+AJ43NV5Rtxtx5+nuLf9kOclIOBRuwwCgnuM0 VK4Mg+svmfczGsffotPe24w= =CcGs -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New voicemail app (supports many interfaces, including Audix)
Are you selling/licensing the new voicemail app or just asking if people want to download it? The reason for asking is if you are selling it I have some thoughts on how voicemail on asterisk can be improved and would like to discuss licensing this to you. Not really working for the next few days till after new year though so email replies will be sporadic. Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Justin Newman Sent: Thursday, 27 December 2007 5:38 PM To: asterisk-users@lists.digium.com Cc: [EMAIL PROTECTED] Subject: [asterisk-users] New voicemail app (supports many interfaces,including Audix) We just completed a new implementation of voicemail for Asterisk. It's much cleaner than Comedian mail and can emulate several voicemail user interfaces, including Audix. It's a great replacement for Audix. All of the sounds/prompts are presently being re-recorded by a professional female voice. If you are interest in the app, let us know at [EMAIL PROTECTED] Justin Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://us.rd.yahoo.com/evt=51734/*http:/tools.search.yahoo.com/newsearc h/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] New voicemail app (supports many interfaces, including Audix)
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Justin Newman wrote: We just completed a new implementation of voicemail for Asterisk. It's much cleaner than Comedian mail and can emulate several voicemail user interfaces, including Audix. It's a great replacement for Audix. All of the sounds/prompts are presently being re-recorded by a professional female voice. If you are interest in the app, let us know at [EMAIL PROTECTED] Also, are you the guy who wrote nvfaxdetect et al? Any chance of an update for 1.4 etc? - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHdDCkDQNt8rg0Kp4RAkKYAJ4v4Y3/unTW9+7F8E3nu0TZvtD8SQCfbwZT VB/vmTfZuwy/W8tNQqReqBU= =OXPQ -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New voicemail app (supports many interfaces, including Audix)
On Fri, Dec 28, 2007 at 12:09:24PM +1300, Matt Riddell wrote: Also, are you the guy who wrote nvfaxdetect et al? Any chance of an update for 1.4 etc? For a version that at least build: http://sourceforge.net/projects/agx-ast-addons (And speaking of new voicemail - what about minivm?) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New voicemail app (supports many interfaces, including Audix)
Yes, I wrote nvfaxdetect and a number of other modules. I don't have any nvfaxdetect updates planned for public release unless someone would like to integrate some of my changes in the GPL version...we could do this though. - Original Message From: Matt Riddell [EMAIL PROTECTED] Justin Newman wrote: We just completed a new implementation of voicemail for Asterisk. It's much cleaner than Comedian mail and can emulate several voicemail user interfaces, including Audix. It's a great replacement for Audix. All of the sounds/prompts are presently being re-recorded by a professional female voice. Also, are you the guy who wrote nvfaxdetect et al? Any chance of an update for 1.4 etc? 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] Gotoiftime help
troxlinux wrote: Verbosity is at least 20 -- Executing [EMAIL PROTECTED]:1] Answer(SIP/sipurafxo-b77038e8, ) in new stack -- Executing [EMAIL PROTECTED]:5] BackGround(SIP/sipurafxo-b77038e8, Well that didn't help any. I'll be working on my system this weekend and I'll see if I can get it to work. I'll let you know. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New voicemail app (supports many interfaces, including Audix)
Licensing your thoughts, do you have a unique patent or a even a patent on an improvement? Aren't you the guy soliciting the user's list for The Next Geewhiz App idea a while ago? Sharks are everywhere. Anyways, this is the Users, soliciting should be done on the Biz list. Thanks, Steve Totaro Dean Collins wrote: Are you selling/licensing the new voicemail app or just asking if people want to download it? The reason for asking is if you are selling it I have some thoughts on how voicemail on asterisk can be improved and would like to discuss licensing this to you. Not really working for the next few days till after new year though so email replies will be sporadic. Cheers, Dean *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Justin Newman *Sent:* Thursday, 27 December 2007 5:38 PM *To:* asterisk-users@lists.digium.com *Cc:* [EMAIL PROTECTED] *Subject:* [asterisk-users] New voicemail app (supports many interfaces,including Audix) We just completed a new implementation of voicemail for Asterisk. It's much cleaner than Comedian mail and can emulate several voicemail user interfaces, including Audix. It's a great replacement for Audix. All of the sounds/prompts are presently being re-recorded by a professional female voice. If you are interest in the app, let us know at [EMAIL PROTECTED] Justin Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://us.rd.yahoo.com/evt=51734/*http:/tools.search.yahoo.com/newsearch/category.php?category=shopping ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New voicemail app (supports many interfaces, including Audix)
So you're saying people like snapanumber, mexuar and other commercially related Asterisk applications cant charge money huh Steve? Maybe this conference call may interest you. http://recordings.talkshoe.com/TC-22622/TS-75263.mp3 Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, 27 December 2007 7:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] New voicemail app (supports many interfaces, including Audix) Licensing your thoughts, do you have a unique patent or a even a patent on an improvement? Aren't you the guy soliciting the user's list for The Next Geewhiz App idea a while ago? Sharks are everywhere. Anyways, this is the Users, soliciting should be done on the Biz list. Thanks, Steve Totaro Dean Collins wrote: Are you selling/licensing the new voicemail app or just asking if people want to download it? The reason for asking is if you are selling it I have some thoughts on how voicemail on asterisk can be improved and would like to discuss licensing this to you. Not really working for the next few days till after new year though so email replies will be sporadic. Cheers, Dean *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Justin Newman *Sent:* Thursday, 27 December 2007 5:38 PM *To:* asterisk-users@lists.digium.com *Cc:* [EMAIL PROTECTED] *Subject:* [asterisk-users] New voicemail app (supports many interfaces,including Audix) We just completed a new implementation of voicemail for Asterisk. It's much cleaner than Comedian mail and can emulate several voicemail user interfaces, including Audix. It's a great replacement for Audix. All of the sounds/prompts are presently being re-recorded by a professional female voice. If you are interest in the app, let us know at [EMAIL PROTECTED] Justin Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://us.rd.yahoo.com/evt=51734/*http:/tools.search.yahoo.com/newsearc h/ category.php?category=shopping ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided 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] CDR
- Original Message From: Steve Murphy [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, 27 December, 2007 5:44:01 PM Subject: Re: [asterisk-users] CDR Greyman-- No real new functionality in 1.4, except a cdr.conf option that lets you control whether you see one-channel cdrs. I haven't been working on CDR's the last few months in favor of other projects that seem a little more urgent. Plus, I have some folks urging me NOT to proceed until some architectural issues are discussed, which might be wise. I have been working on one bug where I did make some substantive changes to how the CDR's are generated, but it is almost certain that these changes will only show up in trunk. I've reached the limit of what I can do in 1.4; it is simply impossible to do anything with CDR's in 1.4 without tearing the very fabric of time and space, and just plain getting everybody upset... at least, those who were not erased from existence by the tear... on a more serious note, the changes are intrusive enough, the behavior changes big enough, that they really don't qualify to be applied to a current release. It's a huge job! My past work was just in the ZAP channel driver code, and because it's so asynch, and all split up into different code, it's really tough to get the right pieces in the right places at the right time in the right way. What this all says is that I'm most likely NOT doing it the right way. And what worries me most is that there might not be any right way. But I'm still new to this, and will get back around to it hopefully fairly soon. murf Hi Steve, Thanks for the update. I agree it's complicated and looks like it does require a look at the design of Asterisk and where CDR's are generated. As you've already documented and lots of us have discovered generating a single CDR for each bridged call is not suitable when CDR's are used for billing and blind and attended transfers are taking place. For any SIP (can't speak for other channels but most likely the same) service providers running Asterisk that are not aware of this problem you will not be getting correct CDR's on blind and attended transfers. Also depending on your dial plan users may be able to send a 302 Redirect response (301 or 302) to an incoming call and get a free outgoing call. This has the potential to cost you money which is very dangerous if any of your users cotton on to it. The easiest way to check your susceptibility is to do call an expensive destination, blind transfer to a free destination and then check the CDRs and pay close attention to the call durations of each CDR. I'll go back to trying to find a way to detect and block dangerous REFER requests at the SIP Proxy before they get to Asterisk. 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
