Re: [asterisk-users] Digium Fax Driver
On Sun, Jun 07, 2009 at 10:15:42PM -0500, Tilghman Lesher wrote: IAXmodem is a completely different ball of wax, and I think you would agree that if the builtin FAX support in spandsp provided excellent support, there never would have been a reason for IAXmodem to be developed. Reminder: SpanDSP was originally GPL. Steve Underwood wrote rx_fax.c and tx_fax.c that were basically thin wrappers for it. They were not so easy to use because: 1. They were out of tree to start with (which made life much more difficult. Certainly before 1.4). 2. The API and ABI of SpanDSP kept changing, and you had to know the right version to use. 3. They weren't much well-behaving Asterisk apps (e.g.: used their own log file) and generally got very little maintinance (also due to (2)). There was much resistance to even adding those apps to asterisk-addons. I guess that merely (2) would have made them a support burden. IAXmodem avoided the issue by using its own copy of SpanDSP. With the release of SpanDSP 0.0.5 things started changing: A. The license was changed to LGPL for the library itself. This allowed using it is Asterisk and FreeSwitch. B. Finally some attention to stable API and ABI. An app_fax.c was written initially in Asterisk-Addons only about at this stage, with an eye on T.38 integration. Shortly after it was moved to Asterisk itself. This was when Asterisk 1.6.0 was at release stages, IIRC. At the time IAXmodem was well-mature. So what you wrote should probably be rephrased as: If the builtin FAX support in *Asterisk* provided excellent support, there never would have been a reason for IAXmodem to be developed. (But still see Lee Howard's comment in the other followup). -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call recording in - out
You should look on the log for when the sox command is called, if the invocation makes sense or not. l. 2009/6/7 Joao Gomes Pereira gomespere...@startel.pt Hello I did as you told me, but the problem remains. Im using Asterisk 1.2.x and this is my config: queues.conf - [general] persistentmembers = no [queue_1] persistentmembers = no monitor-format=wav monitor-join=yes monitor-type=MixMonitor wrapuptime=3 timeout=15 strategy=roundrobin retry=5 queue-youarenext= queue-thereare= queue-thankyou= queue-callswaiting= member = Agent/600 member = Agent/601 agents.conf - [general] persistentagents=no [agents] updatecdr=no custom_beep=beep group=1 wrapuptime=19 ackcall=no musiconhold = music group=1 agent = 600,1234,Jose agent = 601,1234,Maria The calls are recordedbut always produces 2 separated files, with in and out. What could be missing? Do I need to create a line in crontab to mix the 2 files? Thanks regards Joao Pereira Kurian Thayil wrote: Hi, I had similar issue which happened when record option was mentioned in both agents.conf and queues.conf. When I commented the recordagentcalls option in agents.conf, it started to work. Mention the monitor option only in the queues.conf file. Do try. Regards, Kurian Thayil. On Sun, 2009-06-07 at 17:51 +0100, Joao Gomes Pereira wrote: Hello to all I'm trying to record the calls going to my queues, but asterisk creates 2 files, one with the inbound and another with the outbound sound. I know Sox should mix the 2 files automatically in the end, but this isn't happening. I have sox installed in my server. How can I force Sox to mix the files? Here is my config: queues.conf- [general] persistentmembers = no monitor-format=wav monitor-join=yes monitor-type=mixmonitor [queue_1] persistentmembers = no monitor-format=wav monitor-join=yes monitor-type=mixmonitor wrapuptime=3 timeout=15 strategy=roundrobin retry=5 member = Agent/600 member = Agent/601 agents.conf- [general] persistentagents=no [agents] updatecdr=no recordagentcalls=yes recordformat=wav monitor-join=yes savecallsin=/var/www/html/recordings/ custom_beep=beep group=1 wrapuptime=19 ackcall=no group=1 agent = 600,1234,Jose agent = 601,1234,Maria Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Snom, Asterisk and Patton SN1400 - sending bye instead of hold
Hey Everyone, i am using Asterisk 1.4.21.1 with a old Patton SN1400 and some SNOM 300,320,360 Devices. In the combination with asterisk and the patton, there are occuring some strange behaviour, due to the calling and answering everything works good, clear voice, great availability. All the devices have to use ulaw, alaw and slinear is available but never the first choice since i use my asterisk in europe. (slinear is available for debugging supposes) But if a calls comes from or go to the SN1400 and someone tries to HOLD a call, the snoms are sending bye instead of hold, Asterisk plays his MOH until the bye reveives, the snoms doesnt understand this and thinks the caller is still on hold. In the SIP Debug i found some things which i cant handle, so i try to ask you whats going on there : The call comes in, the patton routes it to asterisk and the codec invite starts : --FROM PATTON TO ASTERISK-- Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0x14c (ulaw|alaw|slin|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) The last line is mysterious to me. --ASTERISK IS INVITING MY SNOM AT HOME-- Audio is at [ I P - A S T E R I S K ] port 11576 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x40 (slin) to SDP Adding non-codec 0x1 (telephone-event) to SDP --SNOM IS ANSWERING THE CALL-- Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port [ I P - A N G E R U F E N E R ]:13790 Found audio description format pcmu for ID 0 Found audio description format pcma for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x14c (ulaw|alaw|slin|g729), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) The same as above.. --NOW I PRESS HOLD ON THE SNOM, THE SIP STATEMENT TO THE ASTERISK IS-- --- SIP read from [ I P - A N G E R U F E N E R ]:5060 --- BYE sip:[ TEL. CALLER ]...@[ I P - A S T E R I S K ] SIP/2.0 Via: SIP/2.0/UDP [ I P - A N G E R U F E N E R ]:5060;branch=z9hG4bK-o6cb4olp9iv1;rport From: sip:4...@[ I P - A N G E R U F E N E R ]:5060;line=7anx8ofw;tag=e8yr1936gy To: [ MyName in the Snom ], [ MyName in the Snom ], ;tag=as6fec2de7 Call-ID: 055f1d8f752fcd8b52f0f3b71f89e...@[ MyName in the Snom ].dyndns.org CSeq: 2 BYE Max-Forwards: 70 Contact: sip:4...@[ I P - A N G E R U F E N E R ]:5060;line=7anx8ofw;reg-id=1 User-Agent: snom320/7.3.14 Content-Length: 0 As you can see - a BYE is sent. I tested it out many times, it only occures if a call comes from the patton, only sip calls can greatly be holded and transferred. The whole SIP DEBUG is available here, i dont wanted to post this stuff.. ( http://www.agethen.com/sip-debug-patton-snom.txt ) I would be glad if someone can take a look... Kindly regards, Stefan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium Fax Driver
Tzafrir Cohen wrote: On Sun, Jun 07, 2009 at 10:15:42PM -0500, Tilghman Lesher wrote: IAXmodem is a completely different ball of wax, and I think you would agree that if the builtin FAX support in spandsp provided excellent support, there never would have been a reason for IAXmodem to be developed. Reminder: SpanDSP was originally GPL. Steve Underwood wrote rx_fax.c and tx_fax.c that were basically thin wrappers for it. They were not so easy to use because: 1. They were out of tree to start with (which made life much more difficult. Certainly before 1.4). 2. The API and ABI of SpanDSP kept changing, and you had to know the right version to use. 3. They weren't much well-behaving Asterisk apps (e.g.: used their own log file) and generally got very little maintinance (also due to (2)). There was much resistance to even adding those apps to asterisk-addons. I guess that merely (2) would have made them a support burden. IAXmodem avoided the issue by using its own copy of SpanDSP. With the release of SpanDSP 0.0.5 things started changing: A. The license was changed to LGPL for the library itself. This allowed using it is Asterisk and FreeSwitch. B. Finally some attention to stable API and ABI. An app_fax.c was written initially in Asterisk-Addons only about at this stage, with an eye on T.38 integration. Shortly after it was moved to Asterisk itself. This was when Asterisk 1.6.0 was at release stages, IIRC. At the time IAXmodem was well-mature. So what you wrote should probably be rephrased as: If the builtin FAX support in *Asterisk* provided excellent support, there never would have been a reason for IAXmodem to be developed. (But still see Lee Howard's comment in the other followup). OK, back to the original question: Do they provide the same service? I get the point that the Digium Fax may (or may not) provide V.34. Anything else? sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Push to Talk with Call Drop-Out?
How do you transfer/move an active call to an external number via a dialplan using either the app Dial or Transfer or some alternative, then have Asterisk drop out of the connection. Basically, how can we have asterisk dial another external number, transfer the caller, then disconnect and no longer be required - leaving the transferred and called parties connected. This is for an outbound dialer with all the calls terminating via SIP or IAX2. Basically, we would like to offer our customers the ability to send their customers a call with a pre-recorded message and then give them the option of pressing one to speak to a field office agent. However, some of these calls may get lengthy, so we would prefer to not have to bill them for minute usage by remaining stuck on the call. Any insight into this would be greatly appreciated!___ -- Bandwidth and Colocation Provided 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] T.38 pass-through 488 handling problem
Hi! I have the following problem with Asterisk 1.4.23: ATA w/ T.38 Asterisk ATA w/o T.38 INVITE INVITE ---200OK-- ---200OK-- ACK--- ACK--- INVITE w/T.38- --INVITE w/ T.38-- -488-- --ACK- --BYE- -200-- Asterisk does not forward the 488 back to the caller, but hangs up the callee's call leg. Further, the caller's call leg will not be hung up. Is somebody aware of this problem and a fix? thanks klaus ___ -- Bandwidth and Colocation Provided 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] Timeout when dialing dead peer
A regular Dial(somepeer) to a SIP peer which doesn't reply at all seems to not time out, or at least have a very long time out. We have a set up where we can dial two different peers, a primary and a backup peer. If the first one dies completely, so that no error messages (no ICMP unreachables or anything) are returned, Asterisk does not continue in the dial plan but just gets stuck on that one Dial(). I can of course put a time out in the Dial(), but then one call will turn into two calls if the person at the other end is too slow to answer their phone, so this is not very handy. It is possible that qualify would help, but it is not a very nice answer -- Asterisk's use of SIP OPTIONs is non-standard, and it can impose a significant load on the peer. It would be good if Asterisk would give up after not receving any reply after a configurable interval. /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
[asterisk-users] Best free text to speech..
Hi, i need to use a text to speech in my service. What do think is the best free project? Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk manager login with java not working
I am logging into asterisk manager thru a Java program but not able to login, if i use PHP I am able to login. I have attached my java code with this mail. Can someone step me up to go ahead -- Thank you with regards, Gopal, Echoclient.java Description: Binary data ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Strict Routing and canreinvite
Hello, I want to send Media outside Asterisk server, e.g. between peers. In CLI I see: . [Jun 8 13:13:58] VERBOSE[19112] logger.c: -- Native bridging SIP/5060-b7dc5218 and SIP/prov12-09ad3888 . [Jun 8 13:13:58] DEBUG[19112] chan_sip.c: Strict routing enforced for session 3ad367ee48778d2c523a60e62ae86...@85.113.41.129 And media still goes through Asterisk. Why is that? Why strict routing is enforced? Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions ___ -- Bandwidth and Colocation Provided 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] Timeout when dialing dead peer
Benny Amorsen schrieb: A regular Dial(somepeer) to a SIP peer which doesn't reply at all seems to not time out, or at least have a very long time out. We have a set up where we can dial two different peers, a primary and a backup peer. If the first one dies completely, so that no error messages (no ICMP unreachables or anything) are returned, Asterisk does not continue in the dial plan but just gets stuck on that one Dial(). I can of course put a time out in the Dial(), but then one call will turn into two calls if the person at the other end is too slow to answer their phone, so this is not very handy. It is possible that qualify would help, but it is not a very nice answer -- Asterisk's use of SIP OPTIONs is non-standard, and it can impose a significant load on the peer. What kind of client cant handle one packet per minute without getting a higher load? The interval asterisk sends an Options packet is 60 seconds and the default timeout is 2 s for this packet. So i believe this coudnt be a problem, or do you have a problem with the peer when a second invite arrives during an active call? It would be good if Asterisk would give up after not receving any reply after a configurable interval. What your are searching for is called Sip T1 Timeout and i´ve seen that in asterisk 1.6.1.x you can set this timeout in sip.conf. Iam not sure about changing this in other versions. /Benny best regards steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium Fax Driver
On Sunday 07 June 2009 23:29:30 Lee Howard wrote: I only want to clear up any misrepresentations about possible patent infringements by spandsp to which you alluded. My understanding wasn't that Steve violated any patents, but that he actively avoided certain techniques to avoid conflicting with various patents. That said, hours of use in production do not speak to the amount of testing done. Scrutiny of production use exposure does not constitute testing? Well, I would argue that you cannot possibly test real-world conditions without actually placing the test system into the real-world with real-world use (thus, production). I cannot think of a better way to test software than to eventually put it into real-world production use and then have the developers monitor those systems closely. There's more than a few security holes in the world today because developers (falsely) believed that real-world use constituted testing. No, I'm not saying spandsp has security holes, but I am actively challenging your assertion that real-world use constitutes testing. IAXmodem is a completely different ball of wax, and I think you would agree that if the builtin FAX support in spandsp provided excellent support, there never would have been a reason for IAXmodem to be developed. I'm interested to know how you understand my intent in developing IAXmodem differs from what I recall. I developed IAXmodem because I needed to interface HylaFAX through an Asterisk PBX without purchasing additional hardware (other than the T1 cards that were already involved). At the time I worked for a reseller, the native apps (RxFax and TxFax) failed about 5% of the time, a problem which using IAXmodem and hylafax cleared up to less than 1% problems. We thus deployed IAXmodem to customers who needed incoming fax support, because it simply performed better. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Timeout when dialing dead peer
There is a timeout function in the Dial command. The folks who wrote the command obviously felt that setting a programmatic limit on this would cause somebody a problem. If you expect a reply from your SIP peer in 30 seconds, just do Dial(SIP/peer,30) and the line will disconnect in 30 seconds. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stefan Schmidt Sent: Monday, June 08, 2009 7:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Timeout when dialing dead peer Benny Amorsen schrieb: A regular Dial(somepeer) to a SIP peer which doesn't reply at all seems to not time out, or at least have a very long time out. We have a set up where we can dial two different peers, a primary and a backup peer. If the first one dies completely, so that no error messages (no ICMP unreachables or anything) are returned, Asterisk does not continue in the dial plan but just gets stuck on that one Dial(). I can of course put a time out in the Dial(), but then one call will turn into two calls if the person at the other end is too slow to answer their phone, so this is not very handy. It is possible that qualify would help, but it is not a very nice answer -- Asterisk's use of SIP OPTIONs is non-standard, and it can impose a significant load on the peer. What kind of client cant handle one packet per minute without getting a higher load? The interval asterisk sends an Options packet is 60 seconds and the default timeout is 2 s for this packet. So i believe this coudnt be a problem, or do you have a problem with the peer when a second invite arrives during an active call? It would be good if Asterisk would give up after not receving any reply after a configurable interval. What your are searching for is called Sip T1 Timeout and i´ve seen that in asterisk 1.6.1.x you can set this timeout in sip.conf. Iam not sure about changing this in other versions. /Benny best regards steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided 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] Best free text to speech..
Cepstral and Festival are both Free. In Cepstral, you pay a license fee for the voice you use. In Festival, you tune the mechanical voice the way you want. So if you want Truly free, choose Festival. If you want a Human, Professional voice, Cepstral offers a reasonably priced product. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of equis software Sent: Monday, June 08, 2009 6:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Best free text to speech.. Hi, i need to use a text to speech in my service. What do think is the best free project? Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MeetMe: Mute All Lines Automatically?
I'm considering implementing an Asterisk PBX for conferencing. Before I get started, I wanted to make sure that it supports the features that I need. I plan to use Asterisk as a conference bridge only. I want people to be able to use my conference to listen live to lectures/etc, without having to listen to others in the conference. I'm using the FreePBX web interface, and I can't find any options anywhere. Having the user mute their own line is not going to work, mosty because they *won't* do it. Also, I've used a lot of commercial web-based conferencing services. They all have lots of great features in the web interface, like a list of current participants, muting/unmuting specific lines, manual recording, dropping certain callers, etc. Assuming that Asterisk is capable of all this, is there any web-based GUI available to control meetme? Thanks! -- Christopher Stamper Email: christopherstam...@gmail.com Web: http://tinyurl.com/2ooncg gTalk: http://tinyurl.com/6e359r Skype: cdstamper ___ -- Bandwidth and Colocation Provided 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] Digium Fax Driver
Lee Howard wrote: Tilghman Lesher wrote: On Sunday 07 June 2009 19:39:50 Lee Howard wrote: Tilghman Lesher wrote: What's the use case for the Digium driver? Am I missing something by not using it? While they accomplish the same goal, the commercial driver is based upon a different codebase, Ok. provides support for patented fax protocols, Really? V.34-fax (33,600 bps) is supported? I had understood differently. I would research the patents involved, but I am prohibited by employment contract from exploring patents granted. Due to said employment contract prohibitions you can't tell me whether or not Digium's Fax Application supports V.34-fax (33,600 bps)? My understanding is that there are certain aspects of fax that are still under patent, Yes. Specifically V.34. If my understanding is correct the relevant patents expire in a few years. There are actually 3 aspects of FAXing still under patent protection: - Numerous patents relate to V.34, and the last to expire will still be several years away - There is still one patent in-force related to JBIG compression, which expires next February - The TIFF/FX appears to possibly have Xerox and Adobe's claws in it, but the position is not very clear. The Digium FAX driver is clearly stated to not support V.34. I find this odd, as Commetrex, who supply the FAX engine Digium use, are supposed to have a V.34 engine, and other people (e.g. Pika) say they use it. V.34 would have been a real value add over the free options, as a free option can't provide V.34 for several years. Since Commetrex support JBIG, I assume the Digium FAX driver also does. However, JBIG is not that big a win. Support for it in FAX machines seems patchy. Maybe the makers don't want to pay patent royalties. I intend to add JBIG support to spandsp next year when the last patent expires. TIFF/FX seems to be in limbo, with limited support in day to day usage. and those are provided (along with indemnification) by the commercial driver. Understood. But it was my understanding that V.34-fax was not supported by Digium's Fax Application. And if that's correct, then there are no patents for which indemnification is necessary. That's not to say that a commercial fax driver does not have its place with some customers. I only want to clear up any misrepresentations about possible patent infringements by spandsp to which you alluded. See above. I'm not suggesting that the commercial driver is more reliable, only that it enjoys far more testing. Again, regardless of your knowledge of how much testing goes into your employer's product, I question your ability to know with any degree of certainty as to how much testing has been involved with competing products. I certainly know that *I* have no clue with regards to spandsp other than the testing to which I've been witness. So I am curious to know how you are able to make such assertions. The Digium FAX driver is based on the Commetrex engine, which is widely deployed and presumably robust. How reliable the overall package might be is another matter. For a long time, the limiting factor in FAX reliability with Asterisk has been the inability of Asterisk and/or DAHDI to provide a clean audio stream. In commercial FAX servers using spandsp (either in iaxmodem or on its own) the reliability is mostly limited by Asterisk. Recently, with the launch of the Digium FAX driver, Digium has done some fudging in chan_dahdi to try to mitigate these problems. So far, they only seem to have reduced the issues a bit, and not solve them. That said, hours of use in production do not speak to the amount of testing done. Right, but hours of production use on instrumented servers, which save the audio from every failed call for later analysis can do wonders. We've done this with iaxmodem+HylaFAX and with spandsp in Asterisk and Callweaver systems. Some people have handling hundreds of thousands of FAXes a day, so you can quickly build an interesting library of weird behaviours. It was very time consuming to find and understand all the weird stuff real world equipment throws at you, but we got the unexplained call failures to well below 1% about 3 years ago with iaxmodem, and spandsp is now about there too. Scrutiny of production use exposure does not constitute testing? Well, I would argue that you cannot possibly test real-world conditions without actually placing the test system into the real-world with real-world use (thus, production). I cannot think of a better way to test software than to eventually put it into real-world production use and then have the developers monitor those systems closely. IAXmodem is a completely different ball of wax, and I think you would agree that if the builtin FAX
Re: [asterisk-users] Digium Fax Driver
sean darcy wrote: Tzafrir Cohen wrote: On Sun, Jun 07, 2009 at 10:15:42PM -0500, Tilghman Lesher wrote: IAXmodem is a completely different ball of wax, and I think you would agree that if the builtin FAX support in spandsp provided excellent support, there never would have been a reason for IAXmodem to be developed. Reminder: SpanDSP was originally GPL. Steve Underwood wrote rx_fax.c and tx_fax.c that were basically thin wrappers for it. They were not so easy to use because: 1. They were out of tree to start with (which made life much more difficult. Certainly before 1.4). 2. The API and ABI of SpanDSP kept changing, and you had to know the right version to use. 3. They weren't much well-behaving Asterisk apps (e.g.: used their own log file) and generally got very little maintinance (also due to (2)). There was much resistance to even adding those apps to asterisk-addons. I guess that merely (2) would have made them a support burden. IAXmodem avoided the issue by using its own copy of SpanDSP. With the release of SpanDSP 0.0.5 things started changing: A. The license was changed to LGPL for the library itself. This allowed using it is Asterisk and FreeSwitch. B. Finally some attention to stable API and ABI. An app_fax.c was written initially in Asterisk-Addons only about at this stage, with an eye on T.38 integration. Shortly after it was moved to Asterisk itself. This was when Asterisk 1.6.0 was at release stages, IIRC. At the time IAXmodem was well-mature. So what you wrote should probably be rephrased as: If the builtin FAX support in *Asterisk* provided excellent support, there never would have been a reason for IAXmodem to be developed. (But still see Lee Howard's comment in the other followup). OK, back to the original question: Do they provide the same service? I get the point that the Digium Fax may (or may not) provide V.34. Anything else? I think it gives you JBIG compression, which spandsp can't because of patents. It doesn't give you V.34, which is the really interesting thing Digium could have offered. That's about it. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best free text to speech..
2009/6/8 Danny Nicholas da...@debsinc.com Cepstral and Festival are both “Free”. In Cepstral, you pay a license fee for the voice you use. In Festival, you tune the mechanical voice the way you want. So if you want “Truly free”, choose Festival. If you want a Human, “Professional” voice, Cepstral offers a reasonably priced product. This answer implies you're after an english TTS ... ___ -- Bandwidth and Colocation Provided 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] Best free text to speech..
On Mon, Jun 8, 2009 at 9:02 AM, Danny Nicholasda...@debsinc.com wrote: Cepstral and Festival are both “Free”. In Cepstral, you pay a license fee for the voice you use. In Festival, you tune the mechanical voice the way you want. So if you want “Truly free”, choose Festival. If you want a Human, “Professional” voice, Cepstral offers a reasonably priced product. I've never seen prices for Cepstral, but another commercial product is ATT Natural Voices. http://www.naturalvoices.att.com/ I will say that Festival sounds better now than it did a few years ago. I don't know the exact date, but at some point between when we chose to go with Natural Voices and now, Festival released a new algorithm that sounds dramatically better than it used to. I also really love the Scottish voices that come with Festival. They're improper for my usage, but if you have a business catering to customers where those accents would be appropriate it's a nice product. ___ -- Bandwidth and Colocation Provided 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 VM and Android phone?
Hi, Is anybody picking up emails as attachments on an android phone like the t-mobile G1? I had this working a while ago but since I re-installed my asterisk box to a newer build I am unable to open the attachments, I just get told it can't handle the format? I've been through and tried all wav, wav49, gsm and cant seem to open them. Any suggestions? Mike ___ -- Bandwidth and Colocation Provided 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] Teliax: where's the space in CALLERID(num) from?
On Sat, 2009-06-06 at 21:24 +0200, Philipp Kempgen wrote:
exten = s,n,Set(CALLERID(num)=${IF($[0${CALLERID(num)} 140] ?
${MAINSTUB}${CALLERID(num)}:${MAINNUMBER} )} )
^ ^
remove the trailing spaces
You'll also want to remove any spaces from around the question mark
(after your expression).
--
Jared Smith
Training Manager
Digium, Inc.
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Re: [asterisk-users] Timeout when dialing dead peer
Danny Nicholas schrieb: There is a timeout function in the Dial command. The folks who wrote the command obviously felt that setting a programmatic limit on this would cause somebody a problem. If you expect a reply from your SIP peer in 30 seconds, just do Dial(SIP/peer,30) and the line will disconnect in 30 seconds. which will not work in the situation as benny wrote, when the primary peers doesnt answer to any request coming from asterisk. so you will have an 30 second timeout. what i mean is the Sip internal timeout how long a peer is able to answer to this sip packet, which per default is 30 seconds. if you set the dial timeout lower than this sip timeout you will have a lower waiting time, but as benny said, if the client answer too slow its not handy to use. -Original Message- snip Benny Amorsen schrieb: If the first one dies completely, so that no error messages (no ICMP unreachables or anything) are returned, Asterisk does not continue in the dial plan but just gets stuck on that one Dial(). I can of course put a time out in the Dial(), but then one call will turn into two calls if the person at the other end is too slow to answer their phone, so this is not very handy. /snip i made a mistake in asterisk ver. 1.6.2.b2 you are able to setting the sip timers of your own see the sip.conf sample from this version below: ;--- SIP timers ; These timers are used primarily in INVITE transactions. ; The default for Timer T1 is 500 ms or the measured run-trip time between ; Asterisk and the device if you have qualify=yes for the device. ; ;t1min=100 ; Minimum roundtrip time for messages to monitored hosts ; Defaults to 100 ms ;timert1=500; Default T1 timer ; Defaults to 500 ms or the measured round-trip ; time to a peer (qualify=yes). ;timerb=32000 ; Call setup timer. If a provisional response is not received ; in this amount of time, the call will autocongest ; Defaults to 64*timert1 best regards steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Teliax: where's the space in CALLERID(num) from?
Jared Smith wrote:
On Sat, 2009-06-06 at 21:24 +0200, Philipp Kempgen wrote:
exten = s,n,Set(CALLERID(num)=${IF($[0${CALLERID(num)} 140] ?
${MAINSTUB}${CALLERID(num)}:${MAINNUMBER} )} )
^ ^
remove the trailing spaces
You'll also want to remove any spaces from around the question mark
(after your expression).
Thanks.
Now I have someone in the office, I messed around a bit. The only space
that mattered was the last one. None of the spaces inside ${IF...}
mattered. It makes sense since the parser must see Set(CID=whatever's
in the IF clausethen a space)
Is there a way to test this remotely using originate, or some other
CLI command?
sean
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Re: [asterisk-users] Asterisk VM and Android phone?
Fellow Asterisk Users, I'm trying to marry SugarCRM and Asterisk..perhaps starting from elementary features like a pop-up CRM record upon receipt of inbound call, for starters. Anybody who has successfully done this and beyond? What integration tool are you using? Which CRM are you using? What is the best Asterisk integration tool for SugarCRM? /wai-sun wai...@sqci.biz ___ -- Bandwidth and Colocation Provided 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 VM and Android phone?
2009/6/8 Wai-Sun Chia waisun.c...@gmail.com: Fellow Asterisk Users, I'm trying to marry SugarCRM and Asterisk..perhaps starting from elementary features like a pop-up CRM record upon receipt of inbound call, for starters. Anybody who has successfully done this and beyond? What integration tool are you using? Which CRM are you using? What is the best Asterisk integration tool for SugarCRM? /wai-sun I'm sure you didn't mean to hijack my thread. You would be better posting this to a new thread with appropriate subject line. Also take a look at my blog, http://www.g6phf.co.uk as you may find some answers there. Mike ___ -- Bandwidth and Colocation Provided 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] Best free text to speech..
Witch festival version are you talking about? I need spanish(argentinian) voice... On Mon, Jun 8, 2009 at 10:29 AM, David Backeberg dbackeb...@gmail.comwrote: On Mon, Jun 8, 2009 at 9:02 AM, Danny Nicholasda...@debsinc.com wrote: Cepstral and Festival are both “Free”. In Cepstral, you pay a license fee for the voice you use. In Festival, you tune the mechanical voice the way you want. So if you want “Truly free”, choose Festival. If you want a Human, “Professional” voice, Cepstral offers a reasonably priced product. I've never seen prices for Cepstral, but another commercial product is ATT Natural Voices. http://www.naturalvoices.att.com/ I will say that Festival sounds better now than it did a few years ago. I don't know the exact date, but at some point between when we chose to go with Natural Voices and now, Festival released a new algorithm that sounds dramatically better than it used to. I also really love the Scottish voices that come with Festival. They're improper for my usage, but if you have a business catering to customers where those accents would be appropriate it's a nice product. ___ -- Bandwidth and Colocation Provided 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] MeetMe: Mute All Lines Automatically?
On Mon, Jun 8, 2009 at 9:18 AM, Christopher Stamperchristopherstam...@gmail.com wrote: I'm considering implementing an Asterisk PBX for conferencing. Before I get started, I wanted to make sure that it supports the features that I need. I plan to use Asterisk as a conference bridge only. I want people to be able to use my conference to listen live to lectures/etc, without having to listen to others in the conference. have a look at the documentation here : http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe you want this option 'm' — set monitor only mode (Listen only, no talking) hth ___ -- Bandwidth and Colocation Provided 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] T.38 pass-through 488 handling problem
On Mon, Jun 8, 2009 at 2:06 PM, Klaus Darilionklaus.mailingli...@pernau.at wrote: Hi! I have the following problem with Asterisk 1.4.23: ATA w/ T.38 Asterisk ATA w/o T.38 INVITE INVITE ---200OK-- ---200OK-- ACK--- ACK--- INVITE w/T.38- --INVITE w/ T.38-- -488-- --ACK- --BYE- -200-- Asterisk does not forward the 488 back to the caller, but hangs up the callee's call leg. Further, the caller's call leg will not be hung up. Is somebody aware of this problem and a fix? T.38 passthrough is possible if BOTH devices support T.38, so Asterisk don't have to transcode anything. You could try 1.6 with some gateway app (don't remember if there exists any and in what state), or just write a RxFax which would then generate call with TxFax. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best free text to speech..
On Mon, Jun 8, 2009 at 10:51 AM, equis softwareequissoftw...@gmail.com wrote: Witch festival version are you talking about? I need spanish(argentinian) voice... I don't know whether any free programs do spanish TTS. I can tell you that ATT Natural voices does do TTS en Espanol, and that was part of our reason for choosing it. Although the voice we use is a Mexican accent or maybe kindof pan-Central American? You can play with the application on this demo page. It does French too: http://www.research.att.com/~ttsweb/tts/demo.php ___ -- Bandwidth and Colocation Provided 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] Best free text to speech..
On Mon, 8 Jun 2009, David Backeberg wrote: On Mon, Jun 8, 2009 at 10:51 AM, equis softwareequissoftw...@gmail.com wrote: Witch festival version are you talking about? I need spanish(argentinian) voice... I don't know whether any free programs do spanish TTS. I can tell you that ATT Natural voices does do TTS en Espanol, and that was part of our reason for choosing it. Although the voice we use is a Mexican accent or maybe kindof pan-Central American? You can play with the application on this demo page. It does French too: http://www.research.att.com/~ttsweb/tts/demo.php The quality of TTS these days is truly amazing. May I ask what kind of cost was involved with ATT? Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium Fax Driver
Steve Underwood wrote: I've had a kinda-working-but-not-production-ready SIPmodem for ages, which does allow audio and T.38 from the same HylaFAX system, but I haven't found the time to complete it. Regards, Steve It's good to know that it's not been completely shelved, we are all grateful for your hard work. ___ -- Bandwidth and Colocation Provided 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] Teliax: where's the space in CALLERID(num) from?
sean darcy escribió:
Jared Smith wrote:
On Sat, 2009-06-06 at 21:24 +0200, Philipp Kempgen wrote:
exten = s,n,Set(CALLERID(num)=${IF($[0${CALLERID(num)} 140] ?
${MAINSTUB}${CALLERID(num)}:${MAINNUMBER} )} )
^ ^
remove the trailing spaces
You'll also want to remove any spaces from around the question mark
(after your expression).
Thanks.
Now I have someone in the office, I messed around a bit. The only space
that mattered was the last one. None of the spaces inside ${IF...}
mattered. It makes sense since the parser must see Set(CID=whatever's
in the IF clausethen a space)
Is there a way to test this remotely using originate, or some other
CLI command?
sean
If you have chan_oss (the console channel driver) loaded, you can do a
CLI dial command, for example: dial extens...@context . After you see
what happens with the call, you can hangup it with the CLI hangup
command if the dialplan doesn't hangup it already. Also, if your PC or
server has a well configured sound card and speakers, the console call
would be heard too.
Regards,
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
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Re: [asterisk-users] Asterisk VM and Android phone?
I had the same issue with my Windows Mobile phone for a couple of years. I finally realized that if I had the phone use IMAP instead of POP3, I could open the attachments. No clue why as I received lots of attachments on the phone and they always worked. It was only * attachments that didn't open. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Dent Sent: Monday, June 08, 2009 8:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk VM and Android phone? Hi, Is anybody picking up emails as attachments on an android phone like the t-mobile G1? I had this working a while ago but since I re-installed my asterisk box to a newer build I am unable to open the attachments, I just get told it can't handle the format? I've been through and tried all wav, wav49, gsm and cant seem to open them. Any suggestions? Mike ___ -- Bandwidth and Colocation Provided 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] T.38 pass-through 488 handling problem
Atis Lezdins schrieb: On Mon, Jun 8, 2009 at 2:06 PM, Klaus Darilionklaus.mailingli...@pernau.at wrote: Hi! I have the following problem with Asterisk 1.4.23: ATA w/ T.38 Asterisk ATA w/o T.38 INVITE INVITE ---200OK-- ---200OK-- ACK--- ACK--- INVITE w/T.38- --INVITE w/ T.38-- -488-- --ACK- --BYE- -200-- Asterisk does not forward the 488 back to the caller, but hangs up the callee's call leg. Further, the caller's call leg will not be hung up. Is somebody aware of this problem and a fix? T.38 passthrough is possible if BOTH devices support T.38, so Asterisk don't have to transcode anything. You could try 1.6 with some gateway app (don't remember if there exists any and in what state), or just write a RxFax which would then generate call with TxFax. That's not the problem. Asterisk should just relay back the 488 so that Faxing happens with g.711. regards klaus ___ -- Bandwidth and Colocation Provided 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] T.38 pass-through 488 handling problem
On Mon, Jun 8, 2009 at 7:00 PM, Klaus Darilionklaus.mailingli...@pernau.at wrote: Atis Lezdins schrieb: On Mon, Jun 8, 2009 at 2:06 PM, Klaus Darilionklaus.mailingli...@pernau.at wrote: Hi! I have the following problem with Asterisk 1.4.23: ATA w/ T.38 Asterisk ATA w/o T.38 INVITE INVITE ---200OK-- ---200OK-- ACK--- ACK--- INVITE w/T.38- --INVITE w/ T.38-- -488-- --ACK- --BYE- -200-- Asterisk does not forward the 488 back to the caller, but hangs up the callee's call leg. Further, the caller's call leg will not be hung up. Is somebody aware of this problem and a fix? T.38 passthrough is possible if BOTH devices support T.38, so Asterisk don't have to transcode anything. You could try 1.6 with some gateway app (don't remember if there exists any and in what state), or just write a RxFax which would then generate call with TxFax. That's not the problem. Asterisk should just relay back the 488 so that Faxing happens with g.711. Ok, then You have to look into headers and log, Asterisk should say something.. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT: Grandstream, call pickup, ...
Maybe it's just me, but I get the impression that Grandstream is quite uncooperative. We (and others) have asked them multiple times to make the call- pickup code (**) configurable but either they don't understand the request or they're unwilling to do anything about it. http://forums.grandstream.com/node/2848 http://forums.grandstream.com/node/709 Unfortunately their bug tracker is not public. http://esupport.grandstream.com/support/customerportal/Nologin/index.php?module=Ticketsaction=indexticketid=20090513081505fun=detail In addition they seem to violate sox's license by distributing a modified version of it in binary form only and are not responsive about requests for the source code. That leaves a bad taste in my mouth. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to use Dial G option in AEL
Hi, From Asterisk 1.6.1 embedded doc, Dial app G option is : G(context^exten^pri) - If the call is answered, transfer the calling party to the specified priority and the called party to the specified priority+1. Optionally, an extension, or extension and context may be specified. Otherwise, the current extension is used. You cannot use any additional action post answer options in conjunction with this option. How to safely use this option in AEL ? The issue is how to safely specify priority+1 in extensions.ael ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T.38 pass-through 488 handling problem
Klaus Darilion wrote: Atis Lezdins schrieb: On Mon, Jun 8, 2009 at 2:06 PM, Klaus Darilionklaus.mailingli...@pernau.at wrote: Hi! I have the following problem with Asterisk 1.4.23: ATA w/ T.38 Asterisk ATA w/o T.38 INVITE INVITE ---200OK-- ---200OK-- ACK--- ACK--- INVITE w/T.38- --INVITE w/ T.38-- -488-- --ACK- --BYE- -200-- Asterisk does not forward the 488 back to the caller, but hangs up the callee's call leg. Further, the caller's call leg will not be hung up. Is somebody aware of this problem and a fix? T.38 passthrough is possible if BOTH devices support T.38, so Asterisk don't have to transcode anything. You could try 1.6 with some gateway app (don't remember if there exists any and in what state), or just write a RxFax which would then generate call with TxFax. That's not the problem. Asterisk should just relay back the 488 so that Faxing happens with g.711. There seems to be a common misconception about 488. It represents an irrevocable failure of the call. Once a 488 is sent the call is essentially dead. A number of systems are able to continue beyond a 488, and allow further renegotation to another codec, but that it non-standard behaviour. The correct thing is to offer the options of T.38, u-law and A-law. If the other side can't do T.38, it should accept u-law or A-law. If it says 488, your dead. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best free text to speech..
On Mon, Jun 8, 2009 at 11:08 AM, Jeff LaCoursierej...@jeff.net wrote: The quality of TTS these days is truly amazing. May I ask what kind of cost was involved with ATT? All of that was setup before I worked here. It's possible that at the time ATT won against Cepstral for price, or I'm not sure why we would have chosen it over Cepstral. My understanding is that with the ATT your licensing is based on simultaneous TTS streams in use. ___ -- Bandwidth and Colocation Provided 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: Grandstream, call pickup, ...
Decent product, but their support and development are horrible. I showed them that their SIP over TCP implementation was broken and their reply was use udp -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp Kempgen Sent: Monday, June 08, 2009 11:29 AM To: Asterisk Users Subject: [asterisk-users] OT: Grandstream, call pickup, ... Maybe it's just me, but I get the impression that Grandstream is quite uncooperative. We (and others) have asked them multiple times to make the call- pickup code (**) configurable but either they don't understand the request or they're unwilling to do anything about it. http://forums.grandstream.com/node/2848 http://forums.grandstream.com/node/709 Unfortunately their bug tracker is not public. http://esupport.grandstream.com/support/customerportal/Nologin/index.php?mod ule=Ticketsaction=indexticketid=20090513081505fun=detail In addition they seem to violate sox's license by distributing a modified version of it in binary form only and are not responsive about requests for the source code. That leaves a bad taste in my mouth. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided 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] broken pipe in perl agi
Once again you prove your wisdom. I'm going to look into the AMI think, but
this is a good working solution. My original code was copied from an early
daemon I wrote in PERL, thus the bad problems.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Friday, June 05, 2009 6:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] broken pipe in perl agi
On Fri, 5 Jun 2009, Danny Nicholas wrote:
You're on the right track, Steve but that didn't do it either. Here is
the Perl snippet:
use strict;
use warnings;
my $towatch = $ARGV[0];
my $a=0;
my $retval=undef;
# show hints will get hint information from the dialplan
while ($a1) {
my $cmda = '/usr/sbin/asterisk -rx core show hints|';
Get Trunk Information
my %lines;
my $lineseq=0;
$SIG{'PIPE'} = 'IGNORE';
open (my $trunk_info,$cmda) or print STDOUT Broken pipe\n;
if ($trunk_info) {
while ($trunk_info) {
if ($_ =~ /internal/) {
if ($_ =~ /$towatch/) {
$lines{$lineseq} = $_;
$lineseq++;
}
}
}
close $trunk_info;
}
sleep 2;
for (my $i=0;$i=$lineseq;$i++) {
if ($lines{$i}) {
my $c = unpack(x74 a16, $lines{$i});
$c =~ s/\s//gx;
$retval=1;
print STDOUT SET VARIABLE LINESTAT \$c\ \r\n;
STDIN;
}
}
$a++;
}
# if /var/run/asterisk.ctl is not mod 777, no result so we return a dummy
Idle
if (! $retval) {
my $c = Idle;
print STDOUT SET VARIABLE LINESTAT \$c\ \r\n;
STDIN;
}
exit;
If there is an active call on the extension, it works. If not, the
broken pipe message is returned.
I'm still thinking its a protocol issue.
I couldn't replicate the error on my 1.2 box. I don't use hints, so I read
hint data from a file.
I noticed:
0) You don't use an AGI library
1) You don't turn off I/O buffering
2) You aren't reading the AGI environment
3) You have a sleep in between your 2 loops
4) You have a while loop on $a I don't think is needed
5) You could read Asterisk's output from show hints and process it in a
single loop
6) \r is not needed
7) A space before the request terminator is not needed
I'm not much of a Perl weenie, but I made some changes you're welcome to
use or discard :)
#!/usr/bin/perl
use strict;
use warnings;
# define variables
# show hints will get hint information from the dialplan
my $cmda = '/usr/sbin/asterisk -rx show hints|';
# read hint data from a file for testing
# my $cmda = 'cat /home/sedwards/hints|';
my $towatch = $ARGV[0];
# turn off I/O buffering
$| = 1;
# read the AGI environment
while (STDIN)
{
chomp($_);
last if 0 == length($_);
}
# assume idle
print STDOUT SET VARIABLE LINESTAT \Idle\\n;
STDIN;
# get trunk information
$SIG{'PIPE'} = 'IGNORE';
open (my $trunk_info, $cmda) or exit;
while ($trunk_info)
{
if (($_ =~ /internal/)
($_ =~ /$towatch/))
{
my $c = unpack(x74 a16, $_);
$c =~ s/\s//gx;
print STDOUT SET VARIABLE LINESTAT \$c\\n;
STDIN;
}
}
close $trunk_info;
# (end of hintcheck.agi)
OT, but I think I'm liking AMI more than rx in the new code I'm writing.
Thanks in advance,
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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[asterisk-users] SendText and sipsak
Hi, Following advice in voip-info.org, I could successfully send text to a remote SIP endpoint using sipsak and this command : # sipsak -M -v -s sip:7...@192.168.100.123 sip%3a7...@192.168.100.123 -B Lunch time warning: ignoring -i option when in usrloc mode timeout after 500 ms timeout after 1000 ms timeout after 2000 ms timeout after 4000 ms timeout after 4000 ms timeout after 4000 ms timeout after 4000 ms timeout after 4000 ms timeout after 4000 ms timeout after 4000 ms timeout after 4000 ms *** giving up, no final response after 35621.047 ms Is normal for an endpoint to display a SIP MESSAGE without acking it ? Is there a better way to send a text to a remote end without sipsak ? I tried using .call file but couldn't set autoanswer. Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk crash on DAHDI error: No more room in scheduler
Ran after problem(problem was over weekend, and this was ran 2 days after it started) sorry I forgot about the -v, I was doing the command at home: #dahdi_test Opened pseudo dahdi interface, measuring accuracy... 99.999% 99.996% 99.998% 99.999% 99.998% 99.999% 99.999% 99.999% --- Results after 8 passes --- Best: 99.999 -- Worst: 99.996 -- Average: 99.998283, Difference: 99.998286 Log entries when problem started, note the HDLC Abort: [Jun 6 01:08:51] DEBUG[23599] chan_dahdi.c: Set option AUDIO MODE, value: ON(1) on DAHDI/1-1 [Jun 6 01:08:51] DEBUG[23599] chan_dahdi.c: Not yet hungup... Calling hangup once with icause, and clearing call [Jun 6 01:08:51] DEBUG[23599] chan_dahdi.c: Set option AUDIO MODE, value: OFF(0) on DAHDI/1-1 [Jun 6 01:08:51] VERBOSE[23599] logger.c: -- Hungup 'DAHDI/1-1' [Jun 6 01:11:39] NOTICE[19121] chan_dahdi.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 [Jun 6 01:13:22] ERROR[19121] chan_dahdi.c: No more room in scheduler [Jun 6 01:13:22] ERROR[19121] chan_dahdi.c: Asked to delete sched id -1??? [Jun 6 01:13:22] ERROR[19121] chan_dahdi.c: No more room in scheduler [Jun 6 01:13:22] ERROR[19121] chan_dahdi.c: No more room in scheduler [Jun 6 01:13:22] ERROR[19121] chan_dahdi.c: Asked to delete sched id -1??? [Jun 6 01:13:22] ERROR[19121] chan_dahdi.c: No more room in scheduler [Jun 6 01:13:22] ERROR[19121] chan_dahdi.c: No more room in scheduler [Jun 6 01:13:22] ERROR[19121] chan_dahdi.c: Asked to delete sched id -1??? [Jun 6 01:13:22] ERROR[19121] chan_dahdi.c: No more room in scheduler [Jun 6 01:13:23] ERROR[19121] chan_dahdi.c: No more room in scheduler [Jun 6 01:13:23] ERROR[19121] chan_dahdi.c: Asked to delete sched id -1??? Asterisk was still running but no traffic was going over the T1's (they both should have been filled, or near so) I restarted Asterisk and traffic came pouring in except for T1 #1 channels 1-7 and the whole of T1 #2: *CLI dahdi show channels Chan Extension Context Language MOH Interpret pseudodefault default 1from-pstn default 2from-pstn default 3from-pstn default 4from-pstn default 5from-pstn default 6from-pstn default 7from-pstn default 8 XX4988 from-pstn default 9 XX4988 from-pstn default 10 XX4988 from-pstn default 11 XX4988 from-pstn default 12 XX1587 from-pstn default 13 XX4988 from-pstn default 14 XX4988 from-pstn default 15 XX4988 from-pstn default 16 XX4988 from-pstn default 17 XX4988 from-pstn default 18 XX4988 from-pstn default 19 XX4988 from-pstn default 20 XX4988 from-pstn default 21 XX4988 from-pstn default 22 XX4988 from-pstn default 23 XX4988 from-pstn default 25from-pstn default 26from-pstn default 27from-pstn default 28from-pstn default 29from-pstn default 30from-pstn default 31from-pstn default 32from-pstn default 33from-pstn default 34from-pstn default 35from-pstn default 36from-pstn default 37from-pstn default 38from-pstn default 39from-pstn default 40from-pstn default 41from-pstn default 42from-pstn default 43from-pstn default 44from-pstn default 45from-pstn default 46from-pstn
Re: [asterisk-users] Best free text to speech..
Just out of curiosity, how are you planning to use it? (Reading email, etc?) _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of equis software Sent: Monday, June 08, 2009 7:58 AM To: Asterisk Users List Subject: [asterisk-users] Best free text to speech.. Hi, i need to use a text to speech in my service. What do think is the best free project? Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help with asterisk core dump
Hi to all, I recently upgraded a production machine to asterisk 1.4.25. It seems quite stable but after ~5 days of normal operation it core dumped with this result: (gdb) bt #0 0x00516402 in __kernel_vsyscall () #1 0x005b3d20 in raise () from /lib/libc.so.6 #2 0x005b5631 in abort () from /lib/libc.so.6 #3 0x005ebe6b in __libc_message () from /lib/libc.so.6 #4 0x005f3b16 in _int_free () from /lib/libc.so.6 #5 0x005f7070 in free () from /lib/libc.so.6 #6 0x005e2876 in fclose@@GLIBC_2.1 () from /lib/libc.so.6 #7 0x0809eb2a in filestream_destructor (arg=0xb1f0f200) at file.c:340 #8 0x0806e412 in ao2_ref (user_data=0xb1f0f200, delta=-1) at astobj2.c:229 #9 0x0809c1e9 in ast_closestream (f=0xb1f0f200) at file.c:902 #10 0x00b03422 in local_ast_moh_stop (chan=0xb2150fd0) at res_musiconhold.c:1058 #11 0x00a6e510 in sip_indicate (ast=0xb2150fd0, condition=17, data=0x0, datalen=0) at chan_sip.c:4049 #12 0x08081512 in ast_indicate_data (chan=0xb2150fd0, _condition=17, data=0x0, datalen=0) at channel.c:2530 #13 0x08081728 in ast_indicate (chan=0xb2150fd0, condition=17) at channel.c:2475 #14 0x0097fd33 in agent_new (p=0x950fef0, state=0) at chan_agent.c:1139 #15 0x009837cf in agent_request (type=0xb55fc9b4 Agent, format=2, data=0xb55fc9ba, cause=0xb55fcacc) at chan_agent.c:1469 #16 0x0807d897 in ast_request (type=0xb55fc9b4 Agent, format=2, data=0xb55fc9ba, cause=0xb55fcacc) at channel.c:3203 #17 0x00b62d13 in ring_entry (qe=0xb55fed00, tmp=0x98397f8, busies=0xb55fec44) at app_queue.c:1921 #18 0x00b639c0 in ring_one (qe=0xb55fed00, outgoing=0x9496d60, busies=0xb55fec44) at app_queue.c:2071 #19 0x00b6c670 in try_calling (qe=0xb55fed00, options=value optimized out, announceoverride=0x0, url=0x0, tries=0xb55feea0, noption=0xb55fee9c, agi=0x0) at app_queue.c:2960 #20 0x00b6fdca in queue_exec (chan=0x96e44b8, data=0xb5600f28) at app_queue.c:4083 #21 0x080ca6eb in pbx_extension_helper (c=0x96e44b8, con=0x0, context=0x96e4638 electr-cola-distribucion, exten=0x96e4688 51115, priority=3, label=0x0, callerid=0x958ae58 53654664, action=E_SPAWN) at pbx.c:537 #22 0x080cd0b1 in __ast_pbx_run (c=0x96e44b8) at pbx.c:2320 #23 0x080ce1fe in pbx_thread (data=0x96e44b8) at pbx.c:2636 #24 0x080fdfbb in dummy_start (data=0x9e11c38) at utils.c:856 #25 0x0070446b in start_thread () from /lib/libpthread.so.0 #26 0x0065bdbe in clone () from /lib/libc.so.6 It looks like a very random situation, as this was not a high load moment. Also the asterisk log showed this message in the exact instant of the failure: [Jun 8 13:21:13] ERROR[21601] astobj2.c: refcount -1 on object 0xb1f0f200 I understand that a core dump generated by asterisk compiled with (standard) optimized values is marked as useless information, but IMHO it still helps to know what's failing inside it. I appreciate any input about this, could be this a bug? A library problem? Or a server memory problem? Thanks in advance, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided 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] Best free text to speech..
I need to imlplement an IVR service where customers call and put a telephone number, then I reproduce the name and address. On Mon, Jun 8, 2009 at 3:57 PM, Michelle Dupuis supp...@ocg.ca wrote: Just out of curiosity, how are you planning to use it? (Reading email, etc?) -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *equis software *Sent:* Monday, June 08, 2009 7:58 AM *To:* Asterisk Users List *Subject:* [asterisk-users] Best free text to speech.. Hi, i need to use a text to speech in my service. What do think is the best free project? Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Timeout when dialing dead peer
Stefan Schmidt s...@sil.at writes: What kind of client cant handle one packet per minute without getting a higher load? It isn't a client. It handles thousands of connected devices, so it'll be handling perhaps 50 OPTIONS packets every second if I go the qualify route. What your are searching for is called Sip T1 Timeout and i´ve seen that in asterisk 1.6.1.x you can set this timeout in sip.conf. Iam not sure about changing this in other versions. If you're talking about t1min, AFAIK that only applies to monitored devices, i.e. those with qualify=yes. /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] T.38 pass-through 488 handling problem
Klaus Darilion klaus.mailingli...@pernau.at writes: Asterisk does not forward the 488 back to the caller, but hangs up the callee's call leg. Further, the caller's call leg will not be hung up. Is somebody aware of this problem and a fix? This should be fixed in 1.6.x. At least I had pretty much that scenario break on me in 1.4.2x, and 1.6.0.5 worked. /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] Ekiga, Twinkle and from where to start with open source
It works! :D Thanks CS On Sun, Jun 7, 2009 at 8:57 PM, Philipp Kempgen philipp.kemp...@amooma.dewrote: César Sequeira schrieb: I try to connect Qutecom in my Asterisk Server but without success. What field I need to complete? Username; Password; Realm (asterisk IP Address); Default: asterisk Server (asterisk IP Address); Proxy (asterisk IP address); It's correct? Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bloco de Notas: http://cesarsequeira.wordpress.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] Timeout when dialing dead peer
Benny Amorsen schrieb: Stefan Schmidt s...@sil.at writes: What kind of client cant handle one packet per minute without getting a higher load? It isn't a client. It handles thousands of connected devices, so it'll be handling perhaps 50 OPTIONS packets every second if I go the qualify route. if i understand you right you have one server (peer) where thousands of devices are connected and every device is registered to asterisk, and so every options packet will come from asterisk to this device, right? If you have a sip routing server like ser, the server itself could do a Nat keep alive check, and could drops the invite coming from asterisk if the peer isnt reachable. If these devices arent registered to asterisk why do you think that there will be so much options Packets? if you have one peer this will get only one Options packet per minute. if you just have an rtp routing server or something similar you should have a look at ser / openser/opensip for handling these devices directly. What your are searching for is called Sip T1 Timeout and i´ve seen that in asterisk 1.6.1.x you can set this timeout in sip.conf. Iam not sure about changing this in other versions. If you're talking about t1min, AFAIK that only applies to monitored devices, i.e. those with qualify=yes. /Benny i am talking about t1max which is per rfc definition 64xt1min. Which is normally 32000 milliseconds. If you set this down to 15 seconds the timeout would be half than now, but could cause problems with very slow clients. The qualify options only takes affect on t1min when it set to yes. Then t1min would be set to the average qualify value. As i said i think qualify would be the right solution for you. I have a server running with more than 1600 peers, all with qualify on and notify traffic is around 200 pps in the night with no calls and aroung 6kpps (also with rtp traffic) on high load without taking any affect of the system. Our Ser server have a constant load of 600 pps but that is a proxy build for doing nothing else than routing sip packets. best regards steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with asterisk core dump
Miguel Molina wrote: I recently upgraded a production machine to asterisk 1.4.25. It seems quite stable but after ~5 days of normal operation it core dumped with this result: (gdb) bt #0 0x00516402 in __kernel_vsyscall () #1 0x005b3d20 in raise () from /lib/libc.so.6 #2 0x005b5631 in abort () from /lib/libc.so.6 #3 0x005ebe6b in __libc_message () from /lib/libc.so.6 #4 0x005f3b16 in _int_free () from /lib/libc.so.6 #5 0x005f7070 in free () from /lib/libc.so.6 #6 0x005e2876 in fclose@@GLIBC_2.1 () from /lib/libc.so.6 #7 0x0809eb2a in filestream_destructor (arg=0xb1f0f200) at file.c:340 #8 0x0806e412 in ao2_ref (user_data=0xb1f0f200, delta=-1) at astobj2.c:229 #9 0x0809c1e9 in ast_closestream (f=0xb1f0f200) at file.c:902 #10 0x00b03422 in local_ast_moh_stop (chan=0xb2150fd0) at res_musiconhold.c:1058 --- SNIP --- SNIP --- SNIP --- SNIP --- SNIP --- SNIP --- SNIP --- SNIP --- It looks like a very random situation, as this was not a high load moment. Also the asterisk log showed this message in the exact instant of the failure: [Jun 8 13:21:13] ERROR[21601] astobj2.c: refcount -1 on object 0xb1f0f200 I understand that a core dump generated by asterisk compiled with (standard) optimized values is marked as useless information, but IMHO it still helps to know what's failing inside it. I appreciate any input about this, could be this a bug? A library problem? Or a server memory problem? Miguel, It looks like you are running into an acknowledged bug. There are open issues in the bug tracker for both the 1.4 and 1.6 branches: * https://issues.asterisk.org/view.php?id=15109 * https://issues.asterisk.org/view.php?id=15195 Please create an account and add your information to the bug tracker. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ -- Bandwidth and Colocation Provided 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 manager login with java not working
Gopalakrishnan A.N schrieb: I am logging into asterisk manager thru a Java program but not able to login, if i use PHP I am able to login. I have attached my java code with this mail. Can someone step me up to go ahead What does the manager interface respond? What does the CLI say? Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided 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 manager login with java not working
I would recommend you to use Asterisk-Java library has support for manager, agi, etc. http://asterisk-java.org/ -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp Kempgen Sent: lunes, 08 de junio de 2009 07:47 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk manager login with java not working Gopalakrishnan A.N schrieb: I am logging into asterisk manager thru a Java program but not able to login, if i use PHP I am able to login. I have attached my java code with this mail. Can someone step me up to go ahead What does the manager interface respond? What does the CLI say? Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Se certificó que el correo entrante no contiene virus. Comprobada por AVG - www.avg.es Versión: 8.5.339 / Base de datos de virus: 270.12.57/2163 - Fecha de la versión: 06/08/09 12:30:00 ___ -- Bandwidth and Colocation Provided 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] Help with asterisk core dump
Matthew J. Roth escribió: Miguel Molina wrote: I recently upgraded a production machine to asterisk 1.4.25. It seems quite stable but after ~5 days of normal operation it core dumped with this result: (gdb) bt #0 0x00516402 in __kernel_vsyscall () #1 0x005b3d20 in raise () from /lib/libc.so.6 #2 0x005b5631 in abort () from /lib/libc.so.6 #3 0x005ebe6b in __libc_message () from /lib/libc.so.6 #4 0x005f3b16 in _int_free () from /lib/libc.so.6 #5 0x005f7070 in free () from /lib/libc.so.6 #6 0x005e2876 in fclose@@GLIBC_2.1 () from /lib/libc.so.6 #7 0x0809eb2a in filestream_destructor (arg=0xb1f0f200) at file.c:340 #8 0x0806e412 in ao2_ref (user_data=0xb1f0f200, delta=-1) at astobj2.c:229 #9 0x0809c1e9 in ast_closestream (f=0xb1f0f200) at file.c:902 #10 0x00b03422 in local_ast_moh_stop (chan=0xb2150fd0) at res_musiconhold.c:1058 --- SNIP --- SNIP --- SNIP --- SNIP --- SNIP --- SNIP --- SNIP --- SNIP --- It looks like a very random situation, as this was not a high load moment. Also the asterisk log showed this message in the exact instant of the failure: [Jun 8 13:21:13] ERROR[21601] astobj2.c: refcount -1 on object 0xb1f0f200 I understand that a core dump generated by asterisk compiled with (standard) optimized values is marked as useless information, but IMHO it still helps to know what's failing inside it. I appreciate any input about this, could be this a bug? A library problem? Or a server memory problem? Miguel, It looks like you are running into an acknowledged bug. There are open issues in the bug tracker for both the 1.4 and 1.6 branches: * https://issues.asterisk.org/view.php?id=15109 * https://issues.asterisk.org/view.php?id=15195 Please create an account and add your information to the bug tracker. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer Thank you very much for identifying this known bug. I already have an account, so I will be posting my info on the bugtracker tomorrow. Regarding the trace and the error asterisk throws, [Jun 8 13:21:13] ERROR[21601] astobj2.c: refcount -1 on object 0xb1f0f200 And what astobj2.c comments say on here: 00223/* this case must never happen */ 00224if (current_value 0) 00225 ast_log http://www.asterisk.org/doxygen/1.4/logger_8c.html#0eb07c73aa8c3475ef05c5465d9b5703(LOG_ERROR http://www.asterisk.org/doxygen/1.4/logger_8h.html#91193576ec6eae864eeba838a8c821f5, refcount %d on object %p\n, current_value, user_data http://www.asterisk.org/doxygen/1.4/structastobj2.html#cf53b89bb38cfc4a89c83c743970553c); I think mmichelson is right, quoting him on bug 15123: I think, though, that this particular issue is due to poor refcounting practices present in Asterisk 1.4.22 (which have been fixed already, btw).. So this may be confirming his assumption. Regards, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided 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] MeetMe: Mute All Lines Automatically?
On Mon, Jun 8, 2009 at 10:54 AM, Marc Charbonneau timebandit...@gmail.comwrote: On Mon, Jun 8, 2009 at 9:18 AM, Christopher Stamperchristopherstam...@gmail.com wrote: I'm considering implementing an Asterisk PBX for conferencing. Before I get started, I wanted to make sure that it supports the features that I need. I plan to use Asterisk as a conference bridge only. I want people to be able to use my conference to listen live to lectures/etc, without having to listen to others in the conference. have a look at the documentation here : http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe you want this option 'm' — set monitor only mode (Listen only, no talking) Thanks! Works great. For the record, I also found a web interface for meetme (meetme web control) that does everything I need it to. That is, one I got it to work... ;-) -- Christopher Stamper Email: christopherstam...@gmail.com Web: http://tinyurl.com/2ooncg gTalk: http://tinyurl.com/6e359r Skype: cdstamper ___ -- Bandwidth and Colocation Provided 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] MeetMe: Mute All Lines Automatically?
On Mon, Jun 8, 2009 at 10:54 AM, Marc Charbonneau timebandit...@gmail.comwrote: On Mon, Jun 8, 2009 at 9:18 AM, Christopher Stamperchristopherstam...@gmail.com wrote: I'm considering implementing an Asterisk PBX for conferencing. Before I get started, I wanted to make sure that it supports the features that I need. I plan to use Asterisk as a conference bridge only. I want people to be able to use my conference to listen live to lectures/etc, without having to listen to others in the conference. have a look at the documentation here : http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe you want this option 'm' — set monitor only mode (Listen only, no talking) Thanks! Works great. For the record, I also found a web interface for meetme (meetme web control) that does everything I need it to. That is, one I got it to work... ;-) -- Christopher Stamper Email: christopherstam...@gmail.com Web: http://tinyurl.com/2ooncg gTalk: http://tinyurl.com/6e359r Skype: cdstamper ___ -- Bandwidth and Colocation Provided 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] Achoring MEdia
guys...any opinions on the below? --- On Mon, 6/8/09, Jay Ray jonty...@yahoo.com wrote: From: Jay Ray jonty...@yahoo.com Subject: [asterisk-users] Achoring MEdia To: asterisk-users@lists.digium.com Date: Monday, June 8, 2009, 1:43 AM I have 2 hosts that Asterisk is in between of...and for both I have canreinvite=no - but asterisk still sends re-invite to get out of the media path. Proxy 1 -- Asterisk-- Proxy 2 I want asterisk to anshor media.. In extenstions.conf I have an entry to send calls for say 5551000 to proxy 2 and if I suffix that entry with ||t , asterisk does anchor media However, it is rejecting the REFER sip method by sending a 6xx messageproxy 2 is trying to send the call back out as a transfer and hence it sends a REFERhow can I make it such that media is anchored as wel as the REFER method is accepted by asterisk -Inline Attachment Follows- ___ -- Bandwidth and Colocation Provided 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] alsa no input
I have setup asterisk alsa.conf to read the null device for ALSA and
console/dsp input.
asound.conf is
pcm.nullpcm {
type null
}
alsa.conf has
input_device=plug:nullpcm
Then when I call into the Console/dsp I get very choppy audio.
I dont need and data from the microphone. I just want data out on
console/dsp.
Have I not setup the device properly? why would I be getting choppy
audio from
null pcm device? Is there a better way to accomplish ALSA input for the
console/dsp channel?
Thanks,
Jerry
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[asterisk-users] Understanding Call Handling In Asterisk
Hi, I am a newbie to Asterisk; need help understanding three-way conferencing call-transfer features implemented over standard extensions i.e. on a TDM800P card (4 FXO + 4FXS) In Asterisk I have observed that if an extension is already participating in an active call (e.g. Ext A Ext B communicating): 1. An incoming call to one of these active extensions would be presented with call-wait beeps (e.g. Ext A receives call-wait beeps as Ext C is attempting to call Ext A). 2. The call waiting may be answered by pressing Hook-Flash, placing the previously active call on hold (e.g. C answered; A C communicate; B placed on hold). 3. The calls could be toggled by subsequent Hook-Flash's (e.g. A B communicate; C placed on hold). Yes, this is normal behaviour on pretty much every analogue PBX or telco switch. Queries: 1. If the extension which received call-wait beeps hangs-up then the call waiting/the call placed on hold returns as a new call. I was expecting the call to be transferred (A hangs-up, B C communicate), how could the call be transferred? I expected this feature to be available in Asterisk as this is a very normal feature available on any PBX and used extensively in Call Transfer. When you transfer a call, the person initiating the transfer has to be MAKING a call. Example: Ext A receives a call from Ext B. Ext A wants to transfer the call to Ext C. Ext A puts the first call on hold with a hook flash, dials Ext C, then either waits for the Ext C to answer and announces the transfer (e.g. an attended transfer) OR simply hangs up as soon as the call to Ext C starts ringing (e.g. an un-attended or blind transfer). The behaviour you explain is not something available on any switch that I am aware of, and would be highly problematic if it were. If this feature were available, you could get a circumstance where two people who are calling you end up being bridged together on a call, unknown to you. As a bad example, your wife and your girlfriend end up talking to each other because you hung up while one of them call-waited you while you were talking to the other. The scenario I was expecting was: When Ext. A B are in speech and A is getting call wait beeps from C. Now if Ext. A hangs up, C's call will not be transferred to B but will come as a new call to A. Well if A hangs up after answering C (B on hold), then C's call would be transferred to B when A hangs-up. Another point to be noted is when A B are in speech. Either A could call C by putting B on hold 'or' C could call A present itself as a call-waiting. The maximum loop count will never exceed 2 i.e. at any time you would at-most have one active call one call being held. Hence, either ways if the second call be an incoming or an outgoing transfer shall never occur without the will of transferer ;-). 2. How could the extension that received call-wait beeps initiate a three-way conference with the other extensions (A, B C in three-way conference)? I expected this feature to be available in Asterisk as this is a very normal feature available on any PBX and used extensively in 3-way Call Conference. Again, this is NOT a feature available on any analogue PBX that I am aware of. If it were, this would, again, mean that you may get unwanted parties connected together. With the above example, you answer your girlfriend's call while talking to your wife, and all three of you end up in the same conference. What I was expecting is: 3-way conference would never be intiated by pressing another flash but with a special key sequence (Feature Access Code / Flash + some dtmf digit). Suppose A B were in speech and A is getting call wait beeps from C. Now if A presses flash, the call is toggled i.e. A C are brought in speech and B is placed on hold. Subsequent flashes, would also have a similar behaviour. Well if A dials special key sequence (Feature Access Code / Flash + some dtmf digit), then a 3-way conference would be established. Again this can never happen accidentally. Unfortunately, POTS lines do not handle transfering multiple inbound calls very well (with call waiting). This is not an Asterisk issue, POTS lines were not designed to do anything other than handle a single call at a time. You may be able to handle transfering a call-waited call with DTMF signalling. I am certain someone else on the list will be able to give you a definitive answer on that. ___ -- Bandwidth and Colocation Provided 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] chan_dahdi missing in * 1.6.1.1
What is the magic to compile chan_dahdi.so in asterisk 1.6.1.x? I'm on centos 5.3. Also, asterisk 1.4.25 cannot compile chan_dahdi as well while the previous versions do. What changed or what am i missing? There probably isn't magic. If you post the errors you got during the compile we'll be more likely to be able to tell you what's going wrong. Specifically the stuff you got when you said you cannot compile chan_dahdi.so would be important to post. Ah, there are no errors as such during compile or atleast I didn't seem to notice any. However, I do not see chan_dahdi.so in lib/asterisk/modules/ after asterisk install. Attached are some info and config.log from asterisk 1.6.1.1.. # /opt/dahditools/sbin/dahdi_hardware pci::03:04.0 wanpipe- 1923:0040 Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card # /opt/dahditools/sbin/dahdi_scan [1] active=yes alarms=UNCONFIGURED description=wrtdm Board 1 name=WRTDM/0 manufacturer= devicetype= location= basechan=1 totchans=24 irq=0 type=analog port=1,none port=2,none port=3,FXO port=4,FXO port=5,none port=6,none port=7,none port=8,none port=9,none port=10,none port=11,none port=12,none port=13,none port=14,none port=15,none port=16,none port=17,none port=18,none port=19,none port=20,none port=21,none port=22,none port=23,none port=24,none # cat /proc/dahdi/1 Span 1: WRTDM/0 wrtdm Board 1 (MASTER) 1 WRTDM/0/0 2 WRTDM/0/1 3 WRTDM/0/2 4 WRTDM/0/3 5 WRTDM/0/4 6 WRTDM/0/5 7 WRTDM/0/6 8 WRTDM/0/7 9 WRTDM/0/8 10 WRTDM/0/9 11 WRTDM/0/10 12 WRTDM/0/11 13 WRTDM/0/12 14 WRTDM/0/13 15 WRTDM/0/14 16 WRTDM/0/15 17 WRTDM/0/16 18 WRTDM/0/17 19 WRTDM/0/18 20 WRTDM/0/19 21 WRTDM/0/20 22 WRTDM/0/21 23 WRTDM/0/22 24 WRTDM/0/23 I does not matter if i pass --with-dahdi to ./configure script or not. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
