Re: [asterisk-users] Large Asterisk installations (~10, 000 extensions), preferably at universities
It is very simple take openser(opensips/openser/kamalio) the openser community is great, the project have been here and tested for a years in production, used by the biggest companyes (millions!) of users, it's a carrier grade soft ;) in combination of cdrtool + opensips + mediaproxy you can get 100% billing accuracy. 2008/11/28 Yehavi Bourvine [EMAIL PROTECTED] I did a test yesterday and did 1,000 registrations to Asterisk using SIPP. I did the register test since I am using the realtime DB and asterisk does periodic quesries to it for each registered user. Although Asterisk continued to function as usuall, it was in a steady loop querying the DB for the 1,000 users. OK, you convinced me to look at some front end to it. There are mainly three front ends mentioed here: OpenSer, SipExpress and FreeSwitch. Is there some comparison available which will save me from testing all three of them? Is there one which is more used than the others? (so it has more public QA :-) Thanks! __Yehavi: 2008/11/24 Steve Totaro [EMAIL PROTECTED] Fronting with OpenSER or FS, you should have no problems providing you plan to use SIP extensions. What is critical are the max simultaneous trunks you are going to use. I would go TDM although universities have good bandwidth, and SUPERIOR bandwidth between others. I would think a TDM DS3 or two just to be safe. It should be pretty trivial besides gotchas, like cat3 to the rooms, although channel banks may be an even better solution if phones are already in place. Then you just use SIP when needed or wanted, and Asterisk is simple, although more costly. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) On Fri, Nov 21, 2008 at 6:24 PM, Wilton Helm [EMAIL PROTECTED] wrote: Yet another option is a commercial system with in-house staff. I used to maintain a NEC (NEAX 2400) for many years. I went to factory training and had total responsibility for it. Some manufacturers discourage or prevent this, but others are open to it. There are also 3rd party organizations (such as Source) that can supply parts and even expertise for those going that direction. Whether the result would be higher availability than Asterisk, I don't know. Given I'm both a telco guy and a computer guru (CS degree) I'd probably go the Asterisk route myself, because its open and I would have more control. Wilton and bug fixes than any commercial product sold in the intra-industrial channel ... and they won't charge you a $30,000 license fee for the upgrade. ___ -- 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] force channel hangup
Hi, I have 1 zap channel in my house shared among couple people. If someone dials 911, I want that zap channel to be disconnected right away to make way for the 911 call. I dug through voip-info.org and didn't find much. Any hints? try looking into SoftHangup() http://www.voip-info.org/wiki/view/Asterisk+cmd+SoftHangup cheers -- Daniele Santi .o. [EMAIL PROTECTED] ..o () ascii ribbon campaign Linux User #415108 ooo /\ www.asciiribbon.org ___ -- 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] MixMonitor with non-20ms packets
Hi, MixMonitor saves partial conversation when non-standard voice packet size is set (Asterisk 1.4.18.1). For example, if SIP-peer has alaw:30 then saved file would contain only 67% of total conversation. With alaw:20 MixMonitor saves 100% of conversation. It seems that MixMonitor has hardcoded packets per second or samples per packet values. I did a lot of googling, but found nothing related to this issue. Is it a bug or result of misconfiguration? -- Best regards, Grigoriy Puzankin ___ -- 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] originate problem
Sorry guys, My phonecard (connected to the dial-out channel (Zap-8)) was out of money :-) I'm so embarrased... :-D Let's close this question, I answered it myself :) /Johan -Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För Tzafrir Cohen Skickat: den 27 november 2008 18:07 Till: asterisk-users@lists.digium.com Ämne: Re: [asterisk-users] originate problem On Thu, Nov 27, 2008 at 05:02:17PM +0100, Johan Sandgren wrote: Hi there! Trying to originate and dial a number using Zap-8, used to work, but now it just fails. I enabled all debug I found in the source-code and this is the output from asterisk. Can someone understand something from the debug-output what is wrong and direct me to what the problem might be? The setup is correct, trust me, it worked some hours ago, haven't changed anything. Just dialing again and again to test... sometimes the Zap-8 line does not hangup. But I thought restarting asterisk would hang it up? Maybe it's still off hook. ? What device? What version of Asterisk? Thanks, Johan [Nov 27 16:46:25] DEBUG[907] manager.c: Manager received command 'Originate' [Nov 27 16:46:25] DEBUG[907] chan_zap.c: Using channel 8 [Nov 27 16:46:25] DEBUG[907] chan_zap.c: Dialing '0734414119' [Nov 27 16:46:25] DEBUG[907] chan_zap.c: Deferring dialing... [Nov 27 16:46:25] DEBUG[907] devicestate.c: Notification of state change to be queued on device/channel Zap/8-1 [Nov 27 16:46:25] DEBUG[907] devicestate.c: Notification of state change to be queued on device/channel Zap/8 [Nov 27 16:46:25] DEBUG[877] devicestate.c: No provider found, checking channel drivers for Zap - 8-1 [Nov 27 16:46:25] DEBUG[877] devicestate.c: Changing state for Zap/8-1 - state 0 (Unknown) [Nov 27 16:46:25] DEBUG[877] devicestate.c: No provider found, checking channel drivers for Zap - 8 [Nov 27 16:46:25] DEBUG[877] devicestate.c: Changing state for Zap/8 - state 2 (In use) [Nov 27 16:46:25] DEBUG[902] app_queue.c: Device 'Zap/8-1' changed to state '0' (Unknown) but we don't care because they're not a member of any queue. [Nov 27 16:46:25] DEBUG[902] app_queue.c: Device 'Zap/8' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Nov 27 16:46:26] DEBUG[907] chan_zap.c: Exception on 27, channel 8- this doesn't look good... what does it mean? :-O [Nov 27 16:46:26] DEBUG[907] chan_zap.c: option_debug=100 [Nov 27 16:46:26] DEBUG[907] chan_zap.c: Got event Hook Transition Complete(12) on channel 8 (index 0) [Nov 27 16:46:26] DEBUG[907] chan_zap.c: Sent deferred digit string: T0734414119w [Nov 27 16:46:30] DEBUG[907] chan_zap.c: Exception on 27, channel 8 [Nov 27 16:46:30] DEBUG[907] chan_zap.c: option_debug=100 [Nov 27 16:46:30] DEBUG[907] chan_zap.c: Got event Dial Complete(9) on channel 8 (index 0) [Nov 27 16:46:30] DEBUG[907] chan_zap.c: Enabled echo cancellation on channel 8 The call obivously failed... very strange. Even if I restart asterisk it is still not working... :( [Nov 27 16:46:56] DEBUG[907] channel.c: Hanging up channel 'Zap/8-1' [Nov 27 16:46:56] DEBUG[907] chan_zap.c: zt_hangup(Zap/8-1) [Nov 27 16:46:56] DEBUG[907] chan_zap.c: Hangup: channel: 8 index = 0, normal = 27, callwait = -1, thirdcall = -1 [Nov 27 16:46:56] DEBUG[907] chan_zap.c: disabled echo cancellation on channel 8 [Nov 27 16:46:56] DEBUG[907] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/8-1 [Nov 27 16:46:56] DEBUG[907] chan_zap.c: Updated conferencing on 8, with 0 conference users [Nov 27 16:46:56] VERBOSE[907] logger.c: -- Hungup 'Zap/8-1' ___ Johan Sandgren Svep Design Center AB Phone +46 46 192 722 Mobile +46 70 173 4152 Box 1233, 221 05 Lund, Sweden E-mail [EMAIL PROTECTED] Website www.svep.sehttp://www.svep.se/ ___ -- 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 -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Priority between calls from different queues
Hi! I want to know the way that calls are answer in this case... I have queue1 and queue2, one agent that receive call from both queues. queue1 - call1 queue1 - call2 queue2 - call3 queue2 - call4 In my test the agent answer calls in this order: call1,call3,call2 and call4. I think this must be in this order call1,call2, call3, call4 like a big FIFO. Its ok this behavior? Could I set priority between queues? 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 and S-Bus
Hi everyone, I've built an Asterisk server (1.4.22) on a Debian Etch 4.0 base system (Kernel 2.6.18). I have so far installed AsteriskGUI and the Zaptel, libPRI and mISDN drivers. The hardware is a dual processor, dual core Xeon 2ghz (per core) server with one Digium Wildcard B410P (4 FXS - 4 FXO) and one Beronet BN2S0 ISDN card (1 TE - 1 NT). The all is working well, however I am unsure of how to configure the BN2S0 card port 2 (NT) to provide an S-Bus that additional ISDN phones can connect to as extensions. The phone is powering up OK and Asterisk is dectecting when the phone is taken off hook however I receive an error about Asterisk not knowing how to handle the call. As of yet, I have not configured any additional users in users.conf for the ISDN phone as Im not quite sure how to configure them to use that channel. Has anyone ever tried such a configuration here that might be able to give me a few pointers? Any and all help is much appreciated. Thanks. Kind Regards, Steven Moughan - LAKE Communications, Beech House, Greenhills Road, Dublin 24, IRELAND int. +353 1 4031112 fax. +353 1 452 0826 www.lakecommunications.com This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify [EMAIL PROTECTED] This footnote also confirms that this email message has been scanned for the presence of computer viruses and other security threats. Registered Office: Lake Communications Ltd, Beech House, Greenhills Road, Dublin 24, Ireland. Registered No. 59890 ___ -- 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] [SPAM] - Re: FW: cdr_addon_mysql.so did notregister itselfduringload - Email found in subject
Did you install the MySQL libraries? Debian's command is - apt-get install libmysqlclient15-dev Andy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthias Urlichs Sent: 27 November 2008 16:05 To: asterisk-users@lists.digium.com Subject: [SPAM] - Re: [asterisk-users] FW: cdr_addon_mysql.so did notregister itselfduringload - Email found in subject On Thu, 28 Dec 2006 12:34:46 -0600, Savoy, Kevin - Williston, ND wrote: checking for mysql_init in -lmysqlclient... no What do I need to make that say yes? You need to read config.log and check _why_ the link fails. ___ -- 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] length of field names
I had two definitions in sip.conf like: [geishp64_to_mpgeisjhome1] username=geishp64_to_mpgeisjhome1 secret=26 characters long host=x.y.z.1 [geishp64_to_mpgeisjhome2] username=geishp64_to_mpgeisjhome2 secret=26 characters long host=x.y.z.1 When I had BOTH of the above defines in sip.conf seems like 1.4.21 was getting confused on which one to use. auth didnt match was there error I saw. However, if I remove the second entry everything works as expected. (but I do want two, or more entries for testing). Is there a name length limit on username, secret or [X]? Is it an issue that both connections are to the same IP address or host or are my names to long? I thought everything was 256? Thanks, Jerry ___ -- 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] Priority between calls from different queues
On Fri, Nov 28, 2008 at 1:13 PM, equis software [EMAIL PROTECTED] wrote: Hi! I want to know the way that calls are answer in this case... I have queue1 and queue2, one agent that receive call from both queues. queue1 - call1 queue1 - call2 queue2 - call3 queue2 - call4 In my test the agent answer calls in this order: call1,call3,call2 and call4. I think this must be in this order call1,call2, call3, call4 like a big FIFO. Its ok this behavior? Could I set priority between queues? Hello, Queue has lot of different settings, like wrapuptime, strategy, etc. Also two queues usually don't know about each other, with few exceptions. One of them is shared_lastcall (introduced in Asterisk 1.6.0). There's also weight - it will help to give priority to specific queue if multiple calls are ready to go to agent in different queues. Also, you can give priority to different callers within queue by setting QUEUE_PRIO variable before sending call to queue. You could try to describe why you need two queues and what should be rules to distribute calls - so we can help you with overall architecture. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [EMAIL PROTECTED] 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] [SPAM] - Asterisk and S-Bus - Email found in subject
Have you set port 2 as 'NT' in the mISDN config file (not the Asterisk one)? Also, you will probably need to set it to ptmp. You need to configure them in misdn.conf (the Asterisk one this time). Here's the tail of my misdn.conf (4 x BRI): [trunks] ports = 1,2 ; physical port numbers (as defined in mISDN.conf) context = inbound ; context for incoming calls in extensions.conf msns = * ; accept and process every number that comes in and let my extensions.conf sort it out (easiest way) [extensions] ports = 3,4 ; physical port numbers (as defined in mISDN.conf) context = default ; context for internal users in extensions.conf Ports 1 and 2 are my trunk lines - ports 3 and 4 are my ISDN modems. Hope that helps. Andy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: 28 November 2008 11:54 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [SPAM] - [asterisk-users] Asterisk and S-Bus - Email found in subject Hi everyone, I've built an Asterisk server (1.4.22) on a Debian Etch 4.0 base system (Kernel 2.6.18). I have so far installed AsteriskGUI and the Zaptel, libPRI and mISDN drivers. The hardware is a dual processor, dual core Xeon 2ghz (per core) server with one Digium Wildcard B410P (4 FXS - 4 FXO) and one Beronet BN2S0 ISDN card (1 TE - 1 NT). The all is working well, however I am unsure of how to configure the BN2S0 card port 2 (NT) to provide an S-Bus that additional ISDN phones can connect to as extensions. The phone is powering up OK and Asterisk is dectecting when the phone is taken off hook however I receive an error about Asterisk not knowing how to handle the call. As of yet, I have not configured any additional users in users.conf for the ISDN phone as Im not quite sure how to configure them to use that channel. Has anyone ever tried such a configuration here that might be able to give me a few pointers? Any and all help is much appreciated. Thanks. Kind Regards, Steven Moughan - LAKE Communications, Beech House, Greenhills Road, Dublin 24, IRELAND int. +353 1 4031112 fax. +353 1 452 0826 www.lakecommunications.com This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify [EMAIL PROTECTED] This footnote also confirms that this email message has been scanned for the presence of computer viruses and other security threats. Registered Office: Lake Communications Ltd, Beech House, Greenhills Road, Dublin 24, Ireland. Registered No. 59890 ___ -- 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] Priority between calls from different queues
I saw QUEUE_PRIO but it works inside a queue not between queues. I need to use two queues because their have different settings like max time waiting, max amount of calls in queue and others. Regards On Fri, Nov 28, 2008 at 11:34 AM, Atis Lezdins [EMAIL PROTECTED] wrote: On Fri, Nov 28, 2008 at 1:13 PM, equis software [EMAIL PROTECTED] wrote: Hi! I want to know the way that calls are answer in this case... I have queue1 and queue2, one agent that receive call from both queues. queue1 - call1 queue1 - call2 queue2 - call3 queue2 - call4 In my test the agent answer calls in this order: call1,call3,call2 and call4. I think this must be in this order call1,call2, call3, call4 like a big FIFO. Its ok this behavior? Could I set priority between queues? Hello, Queue has lot of different settings, like wrapuptime, strategy, etc. Also two queues usually don't know about each other, with few exceptions. One of them is shared_lastcall (introduced in Asterisk 1.6.0). There's also weight - it will help to give priority to specific queue if multiple calls are ready to go to agent in different queues. Also, you can give priority to different callers within queue by setting QUEUE_PRIO variable before sending call to queue. You could try to describe why you need two queues and what should be rules to distribute calls - so we can help you with overall architecture. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [EMAIL PROTECTED] 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 ___ -- 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] Anonymous callerid
Max Alex schrieb: I have one issue regarding override callerid when i have anonymous call. I have added PAI in sip header and also set sendrpid = yes in sip.conf but the callerid is not overriding while i am sending call to three digit calling like 911. The caller ID sent to emergency or law enforcement numbers is network-provided not user-provided so you can't override it. Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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] Priority between calls from different queues
One thing you also will run into is listed here: http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf. Here is the interesting part: Note that calls are not offered to queue members whilst the announcement is playing and it is possible for callers to slip ahead in the queue as a result. For example, call 1 arrives and is queued. Call 2 arrives ten seconds later and is queued. After twenty seconds, call 1 is played the periodic announce message. Exactly one second after call 1 starts hearing the message an agent becomes free. Since call 1 is tied up with announcements, call 2 is successfully offered to the agent. Call 1 remains on hold and yet a call which arrived later has been serviced. Basically you can see that if you have announcements played, that could cause your order of answered calls to be not what you expect. Cheers, [cid:image001.jpg@01C95142.5DF134F0] Darrin Henshaw | IT Administrator | MCTS: Exchange 2007 | MCSE 2003 | LPIC Ignition Support Center | www.ignition.bmhttp://www.ignition.bm/ Bermuda (441) 496-4319 | Cayman (345) 947-4357 | Halifax (902) 482-1288 Atlanta | Bermuda | Cayman | Halifax From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of equis software Sent: Friday, November 28, 2008 10:06 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Priority between calls from different queues I saw QUEUE_PRIO but it works inside a queue not between queues. I need to use two queues because their have different settings like max time waiting, max amount of calls in queue and others. Regards On Fri, Nov 28, 2008 at 11:34 AM, Atis Lezdins [EMAIL PROTECTED]mailto:[EMAIL PROTECTED] wrote: On Fri, Nov 28, 2008 at 1:13 PM, equis software [EMAIL PROTECTED]mailto:[EMAIL PROTECTED] wrote: Hi! I want to know the way that calls are answer in this case... I have queue1 and queue2, one agent that receive call from both queues. queue1 - call1 queue1 - call2 queue2 - call3 queue2 - call4 In my test the agent answer calls in this order: call1,call3,call2 and call4. I think this must be in this order call1,call2, call3, call4 like a big FIFO. Its ok this behavior? Could I set priority between queues? Hello, Queue has lot of different settings, like wrapuptime, strategy, etc. Also two queues usually don't know about each other, with few exceptions. One of them is shared_lastcall (introduced in Asterisk 1.6.0). There's also weight - it will help to give priority to specific queue if multiple calls are ready to go to agent in different queues. Also, you can give priority to different callers within queue by setting QUEUE_PRIO variable before sending call to queue. You could try to describe why you need two queues and what should be rules to distribute calls - so we can help you with overall architecture. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [EMAIL PROTECTED]mailto:[EMAIL PROTECTED] 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 This email and its attachments may be confidential and are intended solely for the use of the individual or parties' to whom it is addressed. All comments are solely those of the author and do not necessarily represent those of Ignition. If you are not the intended recipient of this email and its attachments, you must take no action based upon them, nor must you copy or show them to anyone. Please contact the sender if you believe you have received this email in error. Thanks for considering the environmental impact before printing this email. inline: image001.jpg___ -- 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] Priority between calls from different queues
On Fri, Nov 28, 2008 at 4:16 PM, Darrin Henshaw [EMAIL PROTECTED] wrote: One thing you also will run into is listed here: http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf. Here is the interesting part: Note that calls are not offered to queue members whilst the announcement is playing and it is possible for callers to slip ahead in the queue as a result. For example, call 1 arrives and is queued. Call 2 arrives ten seconds later and is queued. After twenty seconds, call 1 is played the periodic announce message. Exactly one second after call 1 starts hearing the message an agent becomes free. Since call 1 is tied up with announcements, call 2 is successfully offered to the agent. Call 1 remains on hold and yet a call which arrived later has been serviced. Basically you can see that if you have announcements played, that could cause your order of answered calls to be not what you expect. With queues there are much more such situation than just this one ;) Cheers, Darrin Henshaw | IT Administrator | MCTS: Exchange 2007 | MCSE 2003 | LPIC Ignition Support Center | www.ignition.bm Bermuda (441) 496-4319 | Cayman (345) 947-4357 | Halifax (902) 482-1288 Atlanta | Bermuda | Cayman | Halifax From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of equis software Sent: Friday, November 28, 2008 10:06 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Priority between calls from different queues I saw QUEUE_PRIO but it works inside a queue not between queues. I need to use two queues because their have different settings like max time waiting, max amount of calls in queue and others. For in-between queues you can use weight. So, if queue1 has more weight than queue2, and agent1 is available (and is in both queues), he will receive call from queue1 (no matter how long other caller waits in queue2). Also, there's wrapuptime. It means - how many seconds agent should not receive call after completing previous queue call. So, if agent receives call from queue1 and it has wrapuptime 10 seconds, then he ends call, he might immediately receive call from queue2 - no matter that queue2 has lower weight or whatever settings. To overcome this, you have to enable shared_lastcall (available since 1.6.0). Regards, Atis Regards On Fri, Nov 28, 2008 at 11:34 AM, Atis Lezdins [EMAIL PROTECTED] wrote: On Fri, Nov 28, 2008 at 1:13 PM, equis software [EMAIL PROTECTED] wrote: Hi! I want to know the way that calls are answer in this case... I have queue1 and queue2, one agent that receive call from both queues. queue1 - call1 queue1 - call2 queue2 - call3 queue2 - call4 In my test the agent answer calls in this order: call1,call3,call2 and call4. I think this must be in this order call1,call2, call3, call4 like a big FIFO. Its ok this behavior? Could I set priority between queues? Hello, Queue has lot of different settings, like wrapuptime, strategy, etc. Also two queues usually don't know about each other, with few exceptions. One of them is shared_lastcall (introduced in Asterisk 1.6.0). There's also weight - it will help to give priority to specific queue if multiple calls are ready to go to agent in different queues. Also, you can give priority to different callers within queue by setting QUEUE_PRIO variable before sending call to queue. You could try to describe why you need two queues and what should be rules to distribute calls - so we can help you with overall architecture. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [EMAIL PROTECTED] 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 This email and its attachments may be confidential and are intended solely for the use of the individual or parties' to whom it is addressed. All comments are solely those of the author and do not necessarily represent those of Ignition. If you are not the intended recipient of this email and its attachments, you must take no action based upon them, nor must you copy or show them to anyone. Please contact the sender if you believe you have received this email in error. Thanks for considering the environmental impact before printing this email. ___ -- 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 -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone:
Re: [asterisk-users] Priority between calls from different queues
I understand, but I don´t have announcements. Regards On Fri, Nov 28, 2008 at 12:16 PM, Darrin Henshaw [EMAIL PROTECTED]wrote: One thing you also will run into is listed here: http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf. Here is the interesting part: Note that calls are not offered to queue members whilst the announcement is playing and it is possible for callers to slip ahead in the queue as a result. For example, call 1 arrives and is queued. Call 2 arrives ten seconds later and is queued. After twenty seconds, call 1 is played the periodic announce message. Exactly one second after call 1 starts hearing the message an agent becomes free. Since call 1 is tied up with announcements, call 2 is successfully offered to the agent. Call 1 remains on hold and yet a call which arrived later has been serviced. Basically you can see that if you have announcements played, that could cause your order of answered calls to be not what you expect. Cheers, [image: logo] Darrin Henshaw |* *IT Administrator | MCTS: Exchange 2007 | MCSE 2003 |LPIC Ignition Support Center* *|* *www.ignition.bm Bermuda (441) 496-4319 | Cayman (345) 947-4357 | Halifax (902) 482-1288 Atlanta | Bermuda | Cayman | Halifax *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *equis software *Sent:* Friday, November 28, 2008 10:06 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Priority between calls from different queues I saw QUEUE_PRIO but it works inside a queue not between queues. I need to use two queues because their have different settings like max time waiting, max amount of calls in queue and others. Regards On Fri, Nov 28, 2008 at 11:34 AM, Atis Lezdins [EMAIL PROTECTED] wrote: On Fri, Nov 28, 2008 at 1:13 PM, equis software [EMAIL PROTECTED] wrote: Hi! I want to know the way that calls are answer in this case... I have queue1 and queue2, one agent that receive call from both queues. queue1 - call1 queue1 - call2 queue2 - call3 queue2 - call4 In my test the agent answer calls in this order: call1,call3,call2 and call4. I think this must be in this order call1,call2, call3, call4 like a big FIFO. Its ok this behavior? Could I set priority between queues? Hello, Queue has lot of different settings, like wrapuptime, strategy, etc. Also two queues usually don't know about each other, with few exceptions. One of them is shared_lastcall (introduced in Asterisk 1.6.0). There's also weight - it will help to give priority to specific queue if multiple calls are ready to go to agent in different queues. Also, you can give priority to different callers within queue by setting QUEUE_PRIO variable before sending call to queue. You could try to describe why you need two queues and what should be rules to distribute calls - so we can help you with overall architecture. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [EMAIL PROTECTED] 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 -- This email and its attachments may be confidential and are intended solely for the use of the individual or parties' to whom it is addressed. All comments are solely those of the author and do not necessarily represent those of Ignition. If you are not the intended recipient of this email and its attachments, you must take no action based upon them, nor must you copy or show them to anyone. Please contact the sender if you believe you have received this email in error. Thanks for considering the environmental impact before printing this email. ___ -- 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 image001.jpg___ -- 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] Priority between calls from different queues
In both queues have the same wrapuptime, there´s not a problem... With weight property I can´t resolve my problem...I want to answer calls of both queues sorted by time, like a big FIFO or like if I had only one queue regards On Fri, Nov 28, 2008 at 12:31 PM, Atis Lezdins [EMAIL PROTECTED] wrote: On Fri, Nov 28, 2008 at 4:16 PM, Darrin Henshaw [EMAIL PROTECTED] wrote: One thing you also will run into is listed here: http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf. Here is the interesting part: Note that calls are not offered to queue members whilst the announcement is playing and it is possible for callers to slip ahead in the queue as a result. For example, call 1 arrives and is queued. Call 2 arrives ten seconds later and is queued. After twenty seconds, call 1 is played the periodic announce message. Exactly one second after call 1 starts hearing the message an agent becomes free. Since call 1 is tied up with announcements, call 2 is successfully offered to the agent. Call 1 remains on hold and yet a call which arrived later has been serviced. Basically you can see that if you have announcements played, that could cause your order of answered calls to be not what you expect. With queues there are much more such situation than just this one ;) Cheers, Darrin Henshaw | IT Administrator | MCTS: Exchange 2007 | MCSE 2003 | LPIC Ignition Support Center | www.ignition.bm Bermuda (441) 496-4319 | Cayman (345) 947-4357 | Halifax (902) 482-1288 Atlanta | Bermuda | Cayman | Halifax From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of equis software Sent: Friday, November 28, 2008 10:06 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Priority between calls from different queues I saw QUEUE_PRIO but it works inside a queue not between queues. I need to use two queues because their have different settings like max time waiting, max amount of calls in queue and others. For in-between queues you can use weight. So, if queue1 has more weight than queue2, and agent1 is available (and is in both queues), he will receive call from queue1 (no matter how long other caller waits in queue2). Also, there's wrapuptime. It means - how many seconds agent should not receive call after completing previous queue call. So, if agent receives call from queue1 and it has wrapuptime 10 seconds, then he ends call, he might immediately receive call from queue2 - no matter that queue2 has lower weight or whatever settings. To overcome this, you have to enable shared_lastcall (available since 1.6.0). Regards, Atis Regards On Fri, Nov 28, 2008 at 11:34 AM, Atis Lezdins [EMAIL PROTECTED] wrote: On Fri, Nov 28, 2008 at 1:13 PM, equis software [EMAIL PROTECTED] wrote: Hi! I want to know the way that calls are answer in this case... I have queue1 and queue2, one agent that receive call from both queues. queue1 - call1 queue1 - call2 queue2 - call3 queue2 - call4 In my test the agent answer calls in this order: call1,call3,call2 and call4. I think this must be in this order call1,call2, call3, call4 like a big FIFO. Its ok this behavior? Could I set priority between queues? Hello, Queue has lot of different settings, like wrapuptime, strategy, etc. Also two queues usually don't know about each other, with few exceptions. One of them is shared_lastcall (introduced in Asterisk 1.6.0). There's also weight - it will help to give priority to specific queue if multiple calls are ready to go to agent in different queues. Also, you can give priority to different callers within queue by setting QUEUE_PRIO variable before sending call to queue. You could try to describe why you need two queues and what should be rules to distribute calls - so we can help you with overall architecture. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [EMAIL PROTECTED] 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 This email and its attachments may be confidential and are intended solely for the use of the individual or parties' to whom it is addressed. All comments are solely those of the author and do not necessarily represent those of Ignition. If you are not the intended recipient of this email and its attachments, you must take no action based upon them, nor must you copy or show them to anyone. Please contact the sender if you believe you have received this email in error.
[asterisk-users] RTCP too short
Dear Sir, I'm running Asterisk 1.4.21.2 on a CentOS machineWhen running asterisk -rv I can see a lot of messages about RTCP too short... -- Remote UNIX connection disconnected [Nov 28 13:33:00] WARNING[24863]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:00] WARNING[19803]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:00] WARNING[19803]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:02] WARNING[29804]: rtp.c:891 ast_rtcp_read: RTCP Read too short Can you let me know how to fix this issue? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Calls drop after a couple of minutes.
I have been encountering a rather hard to debug problem for the last couple of months: * Calls are setup fine. * After a couple of minutes, two way audio becomes one-way and the remote or local party drops out of the call. Setup: * Nokia E71i sip on NAT'd network (multihomed linux box) * Remote asterisk 1.4.21 on Ubuntu on public network * using a Finera/Betamax provider to route calls to PSTN. I initially thought it may be a NAT problem and have checked everything on the NAT gateway/firewall. I see no rejected packets hitting the firewall logs. I'm really at a loss as to what could be causing the calls to drop out for one party so regularly. Any clues where I could look further to debug this would be most useful. local firewall: modprobe ip_conntrack_sip ports=5060 modprobe ip_nat_sip # probably not needed since everything is forwarded: $IPTABLES -A FORWARD -s $INTERNAL_NET -d $ANYWHERE -p udp --dport 5060 -j accept-log # sip remote Asterisk server: $MODPROBE ip_conntrack $MODPROBE ip_conntrack_sip ports=5060 $IPTABLES -A INPUT -s $ANYWHERE -d $PUBLIC_ADDR -p udp --dport 5060 -j accept-log # voip $IPTABLES -A OUTPUT -s $PUBLIC_ADDR -d $ANYWHERE -p udp --sport 5060 -j accept-log # voip $IPTABLES -A OUTPUT -s $PUBLIC_ADDR -d $ANYWHERE -p udp --dport 5060 -j accept-log # voip $IPTABLES -A INPUT -s $ANYWHERE -d $PUBLIC_ADDR -p udp --dport 1:2 -j accept-log # voip $IPTABLES -A OUTPUT -s $PUBLIC_ADDR -d $ANYWHERE -p udp --sport 1:2 -j accept-log # voip sip.conf: [101] callerid=Simon Tennant type=friend username=101 secret=xx host=dynamic reinvite=no canreinvite=no mailbox=101 context=from-internal nat=yes port=5060 qualify=yes insecure=very disallow=all allow=alaw also sip.conf [justvoip.com] type=peer host=sip.justvoip.com fromdomain=sip.justvoip.com progressinband=yes disallow=all allow=alaw ; only alaw works with sip1... nat=no canreinvite=no qualify=yes insecure=port,invite username=imagi-justvoip fromuser=00491785450880 secret= registerattempts=0 ; keep trying to register (normally times out after 10 attempts) context=from-external from rtp.conf rtpstart=19000 rtpend=2 -- Simon Tennant _ fixed: .uk +44 20 7043 6756 .de +49 89 420 955 854 mob: .uk +44 78 5335 6047 .de +49 17 8545 0880 xmpp: [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk SIP security
I was looking at my CLI the other day, and found a lot of those types of messages: NOTICE[2242]: chan_sip.c:14383 handle_request_invite: Call from '' to extension '000452555169' rejected because extension not found. Looking at the IP, it originated from Asia and was clearly an attempt to screw with my Asterisk server. My quick fix was simply to block the IP adress at the firewall level. So that was the end of that. What I don`t get is how the person got that far. How could he attempt to dial extensions (even though he probably was in the default context which has nothing in it) when all my SIP peers are either password protected or linked to a fixed IP. And, more to the point, Call from `` means a call from what exactly? It's not one of my phones, it's not one of my peers ..Shouldn't the lack of a peer be enough to block the would-be hacker from tyring extensions? Any help is appreciate, I clearly don't understand SIP peers. 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] Hints stopped working suddently
Valid question. The problem (hints not working) was reported to me by 3 customers within the same 48 hours. I hadn`t changed anything for a while, but I do remember having removed call-limits on the SIP phonesabout 3 weeks ago. Guess nobody missed hints for a while, hence my incorrect statement about having changed nothing. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: Thursday, November 27, 2008 13:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Hints stopped working suddently Mike, I don't want to be a smart ass, but (as you claimed) if you didn't change anything I've had Asterisk and Polycom phones work perfectly with hints for the last 6 months. Suddently, I realize they've stopped working in the last few days. I haven't changed the configuration in any way. how was it working before ? I really want to know, as there may be something else going on in the background. Julian. Mike wrote: Just to follow-up, because this may one day be found by someone with the same issue, I fixed this: My problem was that my sip peers did not have a call-limit setup. For some (unknown to me) reason, hints only work for peers with a call-limit defined (if using realtime, that would mean something numerical, and not NULL). Mike *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Mike *Sent:* Wednesday, November 26, 2008 11:21 *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Hints stopped working suddently Yes I did. Nothing changes, really. And it all looks good. What I don't get is why the status unavailable appears when the phone is disconnected, but the status inuse doesn't when on a call. That unavailable works fine is some sort of proof that everything is setup properly Mike *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Danny Nicholas *Sent:* Wednesday, November 26, 2008 11:18 *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Hints stopped working suddently Have you tried doing core show hints and sip show peers before and after asterisk restart to see what if anything changes? *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Mike *Sent:* Wednesday, November 26, 2008 10:11 AM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Hints stopped working suddently Not at all, I do everything with vi *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Danny Nicholas *Sent:* Wednesday, November 26, 2008 8:51 *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Hints stopped working suddently Do you use the Asterisk GUI? Changes from it can mess with contexts in the dialplan (extensions.conf) and the hints need to remain in the [internal] context. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Mike *Sent:* Wednesday, November 26, 2008 6:33 AM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* [asterisk-users] Hints stopped working suddently Hello, I've had Asterisk and Polycom phones work perfectly with hints for the last 6 months. Suddently, I realize they've stopped working in the last few days. I haven't changed the configuration in any way. I have hints setup (CLI show hints does show the hints, and they seem correct). But when I do dial using one of the SIP registrations, I don't see those hints being changed in the CLI (at verbose) like I used to. My hints keep on showing idle, even though I am making a call. Making this even weirder, if a phone falls off the grid I do get the subscription become unavailable. It's just the on call hint that does not seem to work. So it seems not to be a firewall/routing issue. I don't think it's the phones, since Asterisk doesn't seem to update it's internal hint (show hints command) when I dial out or get a call. Using Asterisk 1.4.22 and Polycom 3.1.1 firmware (latest). And I restarted asterisk just in case, no help. Regards, * * * * *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 __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email
Re: [asterisk-users] Hints stopped working suddently
On 1.4.22.1 the call-limit is a required parameter for hints to work. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Sent: Friday, November 28, 2008 10:03 AM To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-CommercialDiscussion' Subject: Re: [asterisk-users] Hints stopped working suddently Valid question. The problem (hints not working) was reported to me by 3 customers within the same 48 hours. I hadn`t changed anything for a while, but I do remember having removed call-limits on the SIP phonesabout 3 weeks ago. Guess nobody missed hints for a while, hence my incorrect statement about having changed nothing. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: Thursday, November 27, 2008 13:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Hints stopped working suddently Mike, I don't want to be a smart ass, but (as you claimed) if you didn't change anything I've had Asterisk and Polycom phones work perfectly with hints for the last 6 months. Suddently, I realize they've stopped working in the last few days. I haven't changed the configuration in any way. how was it working before ? I really want to know, as there may be something else going on in the background. Julian. Mike wrote: Just to follow-up, because this may one day be found by someone with the same issue, I fixed this: My problem was that my sip peers did not have a call-limit setup. For some (unknown to me) reason, hints only work for peers with a call-limit defined (if using realtime, that would mean something numerical, and not NULL). Mike *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Mike *Sent:* Wednesday, November 26, 2008 11:21 *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Hints stopped working suddently Yes I did. Nothing changes, really. And it all looks good. What I don't get is why the status unavailable appears when the phone is disconnected, but the status inuse doesn't when on a call. That unavailable works fine is some sort of proof that everything is setup properly. Mike *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Danny Nicholas *Sent:* Wednesday, November 26, 2008 11:18 *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Hints stopped working suddently Have you tried doing core show hints and sip show peers before and after asterisk restart to see what if anything changes? *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Mike *Sent:* Wednesday, November 26, 2008 10:11 AM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Hints stopped working suddently Not at all, I do everything with vi *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Danny Nicholas *Sent:* Wednesday, November 26, 2008 8:51 *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Hints stopped working suddently Do you use the Asterisk GUI? Changes from it can mess with contexts in the dialplan (extensions.conf) and the hints need to remain in the [internal] context. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Mike *Sent:* Wednesday, November 26, 2008 6:33 AM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* [asterisk-users] Hints stopped working suddently Hello, I've had Asterisk and Polycom phones work perfectly with hints for the last 6 months. Suddently, I realize they've stopped working in the last few days. I haven't changed the configuration in any way. I have hints setup (CLI show hints does show the hints, and they seem correct). But when I do dial using one of the SIP registrations, I don't see those hints being changed in the CLI (at verbose) like I used to. My hints keep on showing idle, even though I am making a call. Making this even weirder, if a phone falls off the grid I do get the subscription become unavailable. It's just the on call hint that does not seem to work. So it seems not to be a firewall/routing issue. I don't think it's the phones, since Asterisk doesn't seem to update it's internal hint (show hints command) when I dial out or get a call. Using Asterisk 1.4.22 and Polycom 3.1.1 firmware (latest). And I restarted asterisk just in case, no help. Regards, * * * * *Mike* ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users
[asterisk-users] Friday at 12 Noon ET, the VoIP Users Conference reminder
Hi, As usual, you can get all the dial in information at http://VoipUsersConference.org IRC is on Freenode.net #voip-users-conference join this even if you can't call in. Call via SIP: [EMAIL PROTECTED] (thanks to OnSip.com) Call via PSTN (724) 444-7444 DTMF 22622# 1# or try this: [EMAIL PROTECTED] (thanks to IdeaSIP.com) or to just look up talkshoe server IP: ts.x2z.eu (thanks top me for the DNS record) We start about 15 minutes to the hour with an informal chat. Join us anytime. r ___ -- 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] Priority between calls from different queues
On Fri, Nov 28, 2008 at 4:51 PM, equis software [EMAIL PROTECTED] wrote: In both queues have the same wrapuptime, there´s not a problem... With weight property I can´t resolve my problem...I want to answer calls of both queues sorted by time, like a big FIFO or like if I had only one queue I'm afraid that it's not possible. There will be too much cases when one queue can choose to call agent ignoring another queue. What i meant with wrapuptime - even if it's the same (and you don't use shared_lastcall), second queue won't know that agent has just ended conversation - so it will send call to agent. I guess that there would be some more such race conditions for having free agent. If you really need FIFO, you would have much better luck with having one queue and then thinking how to customize it for different callers. Single instance of Queue is built like FIFO for calls (with bucket of agents). For example - wait time you can specify as argument to Queue(). As for different caller amount, you can assign them to groups and use GROUP_COUNT to determine how many they are in each group. If you need some more differentiation, just ask, and we'll try to give ideas. Oh, btw - you could also try to create one fake agent in queue1 and queue2 (with ringinuse=yes) and use Local channel to send those calls to queue-real where your agents reside. However, i'm not sure that this will work, as queue-real might answer channel, even if you set r option.. not sure is this a problem, but it could be complex :) Regards, Atis regards On Fri, Nov 28, 2008 at 12:31 PM, Atis Lezdins [EMAIL PROTECTED] wrote: On Fri, Nov 28, 2008 at 4:16 PM, Darrin Henshaw [EMAIL PROTECTED] wrote: One thing you also will run into is listed here: http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf. Here is the interesting part: Note that calls are not offered to queue members whilst the announcement is playing and it is possible for callers to slip ahead in the queue as a result. For example, call 1 arrives and is queued. Call 2 arrives ten seconds later and is queued. After twenty seconds, call 1 is played the periodic announce message. Exactly one second after call 1 starts hearing the message an agent becomes free. Since call 1 is tied up with announcements, call 2 is successfully offered to the agent. Call 1 remains on hold and yet a call which arrived later has been serviced. Basically you can see that if you have announcements played, that could cause your order of answered calls to be not what you expect. With queues there are much more such situation than just this one ;) Cheers, Darrin Henshaw | IT Administrator | MCTS: Exchange 2007 | MCSE 2003 | LPIC Ignition Support Center | www.ignition.bm Bermuda (441) 496-4319 | Cayman (345) 947-4357 | Halifax (902) 482-1288 Atlanta | Bermuda | Cayman | Halifax From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of equis software Sent: Friday, November 28, 2008 10:06 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Priority between calls from different queues I saw QUEUE_PRIO but it works inside a queue not between queues. I need to use two queues because their have different settings like max time waiting, max amount of calls in queue and others. For in-between queues you can use weight. So, if queue1 has more weight than queue2, and agent1 is available (and is in both queues), he will receive call from queue1 (no matter how long other caller waits in queue2). Also, there's wrapuptime. It means - how many seconds agent should not receive call after completing previous queue call. So, if agent receives call from queue1 and it has wrapuptime 10 seconds, then he ends call, he might immediately receive call from queue2 - no matter that queue2 has lower weight or whatever settings. To overcome this, you have to enable shared_lastcall (available since 1.6.0). Regards, Atis Regards On Fri, Nov 28, 2008 at 11:34 AM, Atis Lezdins [EMAIL PROTECTED] wrote: On Fri, Nov 28, 2008 at 1:13 PM, equis software [EMAIL PROTECTED] wrote: Hi! I want to know the way that calls are answer in this case... I have queue1 and queue2, one agent that receive call from both queues. queue1 - call1 queue1 - call2 queue2 - call3 queue2 - call4 In my test the agent answer calls in this order: call1,call3,call2 and call4. I think this must be in this order call1,call2, call3, call4 like a big FIFO. Its ok this behavior? Could I set priority between queues? Hello, Queue has lot of different settings, like wrapuptime, strategy, etc. Also two queues usually don't know about each other, with few exceptions. One of them is shared_lastcall (introduced in Asterisk 1.6.0). There's also weight - it will help to give priority to specific queue if multiple calls are
[asterisk-users] Asterisk and multicast RTP
Hi, I would need to bridge a SIP call with a multicast RTP channel. Both sides are receiving and transmitting RTP. Googling, I saw that an app_rtppage, which was in the SVN for a while and its not there anymore. It did, I think, only partly what I need (it sent from SIP to the mcast ... not the other way around), but it was a start. Any idea how to do this? I also could use ser/opensips/openser/kamailio with rtpproxy (does rtpproxy support this? it would in any case be a complex modification, I think). But my current setup is based on asterisk, so I'd rather not move it from there or install new apps. Thanks a bunch! Cesc -- Forwarded message -- From: Cesc Santa [EMAIL PROTECTED] Date: Fri, Nov 28, 2008 at 3:26 PM Subject: Asterisk RTP pager To: [EMAIL PROTECTED] Hi, I came across your RTPpage application and just made me very happy. If I may, some questions. * With which Asterisk versions has it been tested? is it in the official tree? * What I'd like to do is to link this RTPpage with incoming SIP calls ... so that all RTP from SIP is dumped to the multicast RTP and viceversa. Is that possible with this application? Thanks for your time, Cesc ___ -- 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] Priority between calls from different queues
Thanks a lot! Your explanation was very clear. Thanks again. On Fri, Nov 28, 2008 at 2:14 PM, Atis Lezdins [EMAIL PROTECTED] wrote: On Fri, Nov 28, 2008 at 4:51 PM, equis software [EMAIL PROTECTED] wrote: In both queues have the same wrapuptime, there´s not a problem... With weight property I can´t resolve my problem...I want to answer calls of both queues sorted by time, like a big FIFO or like if I had only one queue I'm afraid that it's not possible. There will be too much cases when one queue can choose to call agent ignoring another queue. What i meant with wrapuptime - even if it's the same (and you don't use shared_lastcall), second queue won't know that agent has just ended conversation - so it will send call to agent. I guess that there would be some more such race conditions for having free agent. If you really need FIFO, you would have much better luck with having one queue and then thinking how to customize it for different callers. Single instance of Queue is built like FIFO for calls (with bucket of agents). For example - wait time you can specify as argument to Queue(). As for different caller amount, you can assign them to groups and use GROUP_COUNT to determine how many they are in each group. If you need some more differentiation, just ask, and we'll try to give ideas. Oh, btw - you could also try to create one fake agent in queue1 and queue2 (with ringinuse=yes) and use Local channel to send those calls to queue-real where your agents reside. However, i'm not sure that this will work, as queue-real might answer channel, even if you set r option.. not sure is this a problem, but it could be complex :) Regards, Atis regards On Fri, Nov 28, 2008 at 12:31 PM, Atis Lezdins [EMAIL PROTECTED] wrote: On Fri, Nov 28, 2008 at 4:16 PM, Darrin Henshaw [EMAIL PROTECTED] wrote: One thing you also will run into is listed here: http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf. Here is the interesting part: Note that calls are not offered to queue members whilst the announcement is playing and it is possible for callers to slip ahead in the queue as a result. For example, call 1 arrives and is queued. Call 2 arrives ten seconds later and is queued. After twenty seconds, call 1 is played the periodic announce message. Exactly one second after call 1 starts hearing the message an agent becomes free. Since call 1 is tied up with announcements, call 2 is successfully offered to the agent. Call 1 remains on hold and yet a call which arrived later has been serviced. Basically you can see that if you have announcements played, that could cause your order of answered calls to be not what you expect. With queues there are much more such situation than just this one ;) Cheers, Darrin Henshaw | IT Administrator | MCTS: Exchange 2007 | MCSE 2003 | LPIC Ignition Support Center | www.ignition.bm Bermuda (441) 496-4319 | Cayman (345) 947-4357 | Halifax (902) 482-1288 Atlanta | Bermuda | Cayman | Halifax From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of equis software Sent: Friday, November 28, 2008 10:06 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Priority between calls from different queues I saw QUEUE_PRIO but it works inside a queue not between queues. I need to use two queues because their have different settings like max time waiting, max amount of calls in queue and others. For in-between queues you can use weight. So, if queue1 has more weight than queue2, and agent1 is available (and is in both queues), he will receive call from queue1 (no matter how long other caller waits in queue2). Also, there's wrapuptime. It means - how many seconds agent should not receive call after completing previous queue call. So, if agent receives call from queue1 and it has wrapuptime 10 seconds, then he ends call, he might immediately receive call from queue2 - no matter that queue2 has lower weight or whatever settings. To overcome this, you have to enable shared_lastcall (available since 1.6.0). Regards, Atis Regards On Fri, Nov 28, 2008 at 11:34 AM, Atis Lezdins [EMAIL PROTECTED] wrote: On Fri, Nov 28, 2008 at 1:13 PM, equis software [EMAIL PROTECTED] wrote: Hi! I want to know the way that calls are answer in this case... I have queue1 and queue2, one agent that receive call from both queues. queue1 - call1 queue1 - call2 queue2 - call3 queue2 - call4 In my test the agent answer calls in this order: call1,call3,call2 and call4. I think this must be in this order call1,call2, call3, call4 like a big FIFO. Its ok this behavior? Could I set priority between queues? Hello, Queue has lot of
Re: [asterisk-users] RTCP too short
I get this all the time. Still haven't found a solution but it doesnt seem to affect call quality or server performance. I think there's a way to disable the message, but I lost the link. :( -Jon - Original Message - From: michel freiha To: Asterisk Users Mailing List - Non-Commercial Discussion ; [EMAIL PROTECTED] Sent: Friday, November 28, 2008 10:07 AM Subject: [asterisk-users] RTCP too short Dear Sir, I'm running Asterisk 1.4.21.2 on a CentOS machineWhen running asterisk -rv I can see a lot of messages about RTCP too short... -- Remote UNIX connection disconnected [Nov 28 13:33:00] WARNING[24863]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:00] WARNING[19803]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:00] WARNING[19803]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:02] WARNING[29804]: rtp.c:891 ast_rtcp_read: RTCP Read too short Can you let me know how to fix this issue? Regards -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTCP too short
The quick answer is that your realtime isn't transmitting full frames. This message occurs when the number of bytes from the frame read isn't divisible by 4. Changing the rtpchecksums in rtp.conf might correct this. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Weisman Sent: Friday, November 28, 2008 11:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RTCP too short I get this all the time. Still haven't found a solution but it doesnt seem to affect call quality or server performance. I think there's a way to disable the message, but I lost the link. :( -Jon - Original Message - From: michel freiha mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - mailto:asterisk-users@lists.digium.com Non-Commercial Discussion ; [EMAIL PROTECTED] Sent: Friday, November 28, 2008 10:07 AM Subject: [asterisk-users] RTCP too short Dear Sir, I'm running Asterisk 1.4.21.2 on a CentOS machineWhen running asterisk -rv I can see a lot of messages about RTCP too short... -- Remote UNIX connection disconnected [Nov 28 13:33:00] WARNING[24863]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:00] WARNING[19803]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:00] WARNING[19803]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:02] WARNING[29804]: rtp.c:891 ast_rtcp_read: RTCP Read too short Can you let me know how to fix this issue? Regards _ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTCP too short
realtime? I'm using static config files, no realtime. - Original Message - From: Danny Nicholas To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Friday, November 28, 2008 1:02 PM Subject: Re: [asterisk-users] RTCP too short The quick answer is that your realtime isn't transmitting full frames. This message occurs when the number of bytes from the frame read isn't divisible by 4. Changing the rtpchecksums in rtp.conf might correct this. -- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Weisman Sent: Friday, November 28, 2008 11:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RTCP too short I get this all the time. Still haven't found a solution but it doesnt seem to affect call quality or server performance. I think there's a way to disable the message, but I lost the link. :( -Jon - Original Message - From: michel freiha To: Asterisk Users Mailing List - Non-Commercial Discussion ; [EMAIL PROTECTED] Sent: Friday, November 28, 2008 10:07 AM Subject: [asterisk-users] RTCP too short Dear Sir, I'm running Asterisk 1.4.21.2 on a CentOS machineWhen running asterisk -rv I can see a lot of messages about RTCP too short... -- Remote UNIX connection disconnected [Nov 28 13:33:00] WARNING[24863]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:00] WARNING[19803]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:00] WARNING[19803]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:02] WARNING[29804]: rtp.c:891 ast_rtcp_read: RTCP Read too short Can you let me know how to fix this issue? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ -- 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] RTCP too short
Double check your config files. Rtp.c is a real-time component, so you're getting a phantom call to this routine (possibly from CDR?) _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Weisman Sent: Friday, November 28, 2008 12:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RTCP too short realtime? I'm using static config files, no realtime. - Original Message - From: Danny Nicholas mailto:[EMAIL PROTECTED] To: 'Asterisk Users Mailing List - mailto:asterisk-users@lists.digium.com Non-Commercial Discussion' Sent: Friday, November 28, 2008 1:02 PM Subject: Re: [asterisk-users] RTCP too short The quick answer is that your realtime isn't transmitting full frames. This message occurs when the number of bytes from the frame read isn't divisible by 4. Changing the rtpchecksums in rtp.conf might correct this. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Weisman Sent: Friday, November 28, 2008 11:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RTCP too short I get this all the time. Still haven't found a solution but it doesnt seem to affect call quality or server performance. I think there's a way to disable the message, but I lost the link. :( -Jon - Original Message - From: michel freiha mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - mailto:asterisk-users@lists.digium.com Non-Commercial Discussion ; [EMAIL PROTECTED] Sent: Friday, November 28, 2008 10:07 AM Subject: [asterisk-users] RTCP too short Dear Sir, I'm running Asterisk 1.4.21.2 on a CentOS machineWhen running asterisk -rv I can see a lot of messages about RTCP too short... -- Remote UNIX connection disconnected [Nov 28 13:33:00] WARNING[24863]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:00] WARNING[19803]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:00] WARNING[19803]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:02] WARNING[29804]: rtp.c:891 ast_rtcp_read: RTCP Read too short Can you let me know how to fix this issue? Regards _ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] RTCP too short
Do you have Grandstream phones? I noticed a similar issue last year with Grandstream GXP2000 phones. The phone was sending an empty RTP packet for the keepalive whilst on mute. I reported a bug to Grandstream but nothing happened. regards, Drew Jon Weisman wrote: I get this all the time. Still haven't found a solution but it doesnt seem to affect call quality or server performance. I think there's a way to disable the message, but I lost the link. :( -Jon - Original Message - *From:* michel freiha mailto:[EMAIL PROTECTED] *To:* Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com ; [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] *Sent:* Friday, November 28, 2008 10:07 AM *Subject:* [asterisk-users] RTCP too short Dear Sir, I'm running Asterisk 1.4.21.2 http://1.4.21.2 on a CentOS machineWhen running asterisk -rv I can see a lot of messages about RTCP too short... -- Remote UNIX connection disconnected [Nov 28 13:33:00] WARNING[24863]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:00] WARNING[19803]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:00] WARNING[19803]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:02] WARNING[29804]: rtp.c:891 ast_rtcp_read: RTCP Read too short Can you let me know how to fix this issue? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] How to disable trunk from the cli?
Hi, I need to be able to unable and disable iax2 trunks from the cli? Is there a way to do it if so how? Sincerely, Robert Augustyn ___ -- 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 and avaya SIP trunking
Hi Shaun and Robb, I tried the Avaya IP small Office with Lucent Analog phones. it worked fine on hang ups - i think it is my old analog phone is the root cause. I have only one major issue now. I am not getting the Caller ID Name and Caller ID number from avaya to asterisk. Can you provide me your valuable input. This is what i have. When i configured the SIP trunk, Under SIP line - i had Primary authentication name = avayanew Primary authentication Password = avayanew and also under the SIP URI: Local URI, Contact and Display name - i had selected use authentication name for successful calls, but as said caller ID is not passed through asterisk. when i try the use user data in there, i get TTel: the problem i had before and cannot make/receive calls. Please advise Thanks as always Regards Krishna On Mon, Nov 10, 2008 at 6:57 PM, Shaun Ewing [EMAIL PROTECTED] wrote: On Tue, Nov 11, 2008 at 4:56 AM, Krishna Sumanth Chava [EMAIL PROTECTED] wrote: HI Shaun and Robb, Thanks for the assistance. I was able to force the codecs and had avaya talk in the right way. Also addressed the DTMF issues. Glad to hear it. I seem to be having issues with asterisk and avaya not detecting Hang ups. i am using the Analog phones connected to the POTS ports on the IP Office. I will try connecting the avaya Analog and Avaya IP Phone to IP Office and see if that makes any difference. What does SSA show when one end has hung up? If it still shows the call as active, then a disconnect signal has gone missing. I've never experienced this problem, but then again the only thing we use the POTS ports for is faxing and this is forced to use our PRI circuits. All of our handsets including conference room phones are IP. -Shaun ___ -- 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] How to disable trunk from the cli?
Iax2 provision would seem to be a harsh but simple way to do it. Iax2 provision and iax2 prune seem like kinder candidates, but I haven't gotten into the iax2 branch of * yet. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Augustyn Sent: Friday, November 28, 2008 1:07 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How to disable trunk from the cli? Hi, I need to be able to unable and disable iax2 trunks from the cli? Is there a way to do it if so how? Sincerely, Robert Augustyn ___ -- 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 SIP security
On Fri, Nov 28, 2008 at 11:00 AM, Mike [EMAIL PROTECTED] wrote: I was looking at my CLI the other day, and found a lot of those types of messages: NOTICE[2242]: chan_sip.c:14383 handle_request_invite: Call from '' to extension '000452555169' rejected because extension not found. Looking at the IP, it originated from Asia and was clearly an attempt to screw with my Asterisk server. My quick fix was simply to block the IP adress at the firewall level. So that was the end of that. What I don`t get is how the person got that far. How could he attempt to dial extensions (even though he probably was in the default context which has nothing in it) when all my SIP peers are either password protected or linked to a fixed IP. And, more to the point, Call from `` means a call from what exactly? It's not one of my phones, it's not one of my peers…..Shouldn't the lack of a peer be enough to block the would-be hacker from tyring extensions? Any help is appreciate, I clearly don't understand SIP peers. Mike I think if you remove context from the [general] section, you would not see these messages. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] RTCP too short
yea definitly not using realtime. i am logging cdr to mysql, could that be it? - Original Message - From: Danny Nicholas To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Friday, November 28, 2008 1:22 PM Subject: Re: [asterisk-users] RTCP too short Double check your config files. Rtp.c is a real-time component, so you're getting a phantom call to this routine (possibly from CDR?) -- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Weisman Sent: Friday, November 28, 2008 12:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RTCP too short realtime? I'm using static config files, no realtime. - Original Message - From: Danny Nicholas To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Friday, November 28, 2008 1:02 PM Subject: Re: [asterisk-users] RTCP too short The quick answer is that your realtime isn't transmitting full frames. This message occurs when the number of bytes from the frame read isn't divisible by 4. Changing the rtpchecksums in rtp.conf might correct this. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Weisman Sent: Friday, November 28, 2008 11:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RTCP too short I get this all the time. Still haven't found a solution but it doesnt seem to affect call quality or server performance. I think there's a way to disable the message, but I lost the link. :( -Jon - Original Message - From: michel freiha To: Asterisk Users Mailing List - Non-Commercial Discussion ; [EMAIL PROTECTED] Sent: Friday, November 28, 2008 10:07 AM Subject: [asterisk-users] RTCP too short Dear Sir, I'm running Asterisk 1.4.21.2 on a CentOS machineWhen running asterisk -rv I can see a lot of messages about RTCP too short... -- Remote UNIX connection disconnected [Nov 28 13:33:00] WARNING[24863]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:00] WARNING[19803]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:00] WARNING[19803]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:02] WARNING[29804]: rtp.c:891 ast_rtcp_read: RTCP Read too short Can you let me know how to fix this issue? Regards -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ___ -- 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] RTCP too short
nope, no grandstreams...already learned not to use them the hard way - Original Message - From: Drew Gibson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, November 28, 2008 1:52 PM Subject: Re: [asterisk-users] RTCP too short Do you have Grandstream phones? I noticed a similar issue last year with Grandstream GXP2000 phones. The phone was sending an empty RTP packet for the keepalive whilst on mute. I reported a bug to Grandstream but nothing happened. regards, Drew Jon Weisman wrote: I get this all the time. Still haven't found a solution but it doesnt seem to affect call quality or server performance. I think there's a way to disable the message, but I lost the link. :( -Jon - Original Message - *From:* michel freiha mailto:[EMAIL PROTECTED] *To:* Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com ; [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] *Sent:* Friday, November 28, 2008 10:07 AM *Subject:* [asterisk-users] RTCP too short Dear Sir, I'm running Asterisk 1.4.21.2 http://1.4.21.2 on a CentOS machineWhen running asterisk -rv I can see a lot of messages about RTCP too short... -- Remote UNIX connection disconnected [Nov 28 13:33:00] WARNING[24863]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:00] WARNING[19803]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:00] WARNING[19803]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:02] WARNING[29804]: rtp.c:891 ast_rtcp_read: RTCP Read too short Can you let me know how to fix this issue? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTCP too short
Could be. My best guess is that it's either the keepalive issue from your hardphone or that using mysql is jumping into the rtp logic. If you made CDR go to a text file and the problem did not go away, that would isolate it as a hardphone issue IMO. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Weisman Sent: Friday, November 28, 2008 2:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RTCP too short yea definitly not using realtime. i am logging cdr to mysql, could that be it? - Original Message - From: Danny Nicholas mailto:[EMAIL PROTECTED] To: 'Asterisk Users Mailing List - mailto:asterisk-users@lists.digium.com Non-Commercial Discussion' Sent: Friday, November 28, 2008 1:22 PM Subject: Re: [asterisk-users] RTCP too short Double check your config files. Rtp.c is a real-time component, so you're getting a phantom call to this routine (possibly from CDR?) _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Weisman Sent: Friday, November 28, 2008 12:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RTCP too short realtime? I'm using static config files, no realtime. - Original Message - From: Danny Nicholas mailto:[EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial mailto:asterisk-users@lists.digium.com Discussion' Sent: Friday, November 28, 2008 1:02 PM Subject: Re: [asterisk-users] RTCP too short The quick answer is that your realtime isn't transmitting full frames. This message occurs when the number of bytes from the frame read isn't divisible by 4. Changing the rtpchecksums in rtp.conf might correct this. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Weisman Sent: Friday, November 28, 2008 11:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RTCP too short I get this all the time. Still haven't found a solution but it doesnt seem to affect call quality or server performance. I think there's a way to disable the message, but I lost the link. :( -Jon - Original Message - From: michel freiha mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - mailto:asterisk-users@lists.digium.com Non-Commercial Discussion ; [EMAIL PROTECTED] Sent: Friday, November 28, 2008 10:07 AM Subject: [asterisk-users] RTCP too short Dear Sir, I'm running Asterisk 1.4.21.2 on a CentOS machineWhen running asterisk -rv I can see a lot of messages about RTCP too short... -- Remote UNIX connection disconnected [Nov 28 13:33:00] WARNING[24863]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:00] WARNING[19803]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:00] WARNING[19803]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:02] WARNING[29804]: rtp.c:891 ast_rtcp_read: RTCP Read too short Can you let me know how to fix this issue? Regards _ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ ___ -- 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] Any 1.6 SendFAX example ?
On Thursday 27 November 2008 05:03:00 Olivier wrote: Hi, Do you have any example showing how to use SendFAX ? I can see several examples of ReceiveFAX but not a single one showing SendFAX. i'm working on a script to incorporate e-mail - fax gatewaying with asterisk using programs that are already available in linux. there are simple examples here: http://messinet.com/viewvc/asterisk-fax-gw/trunk/ -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] force channel hangup
On Thursday 27 November 2008 20:25:49 Kelvin Chan wrote: Hi guys, I have 1 zap channel in my house shared among couple people. If someone dials 911, I want that zap channel to be disconnected right away to make way for the 911 call. I dug through voip-info.org and didn't find much. Any hints? i use this: http://messinet.com/index.php?page_name=Asteriskwikipage=Asteriske911 -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] force channel hangup
Why wouldn't this work? exten = _911,1,Hangup(Zap/1) exten = _911,2,Dial(Zap/1/ww911,60) exten = _911,3,Hangup() -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Messina Sent: Friday, November 28, 2008 3:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] force channel hangup On Thursday 27 November 2008 20:25:49 Kelvin Chan wrote: Hi guys, I have 1 zap channel in my house shared among couple people. If someone dials 911, I want that zap channel to be disconnected right away to make way for the 911 call. I dug through voip-info.org and didn't find much. Any hints? i use this: http://messinet.com/index.php?page_name=Asteriskwikipage=Asteriske911 -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E ___ -- 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] force channel hangup
Because hangup (and other behavioural) directives can only be addressed to a particular instance of a channel use, i.e. Technology/channel-uniqueID. The latter are not addressable from the dial plan except implicitly. Danny Nicholas wrote: Why wouldn't this work? exten = _911,1,Hangup(Zap/1) exten = _911,2,Dial(Zap/1/ww911,60) exten = _911,3,Hangup() -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Messina Sent: Friday, November 28, 2008 3:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] force channel hangup On Thursday 27 November 2008 20:25:49 Kelvin Chan wrote: Hi guys, I have 1 zap channel in my house shared among couple people. If someone dials 911, I want that zap channel to be disconnected right away to make way for the 911 call. I dug through voip-info.org and didn't find much. Any hints? i use this: http://messinet.com/index.php?page_name=Asteriskwikipage=Asteriske911 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] force channel hangup
Right you are, Alex. How about (CLI) Zap restart? I was thinking zap
destroy channel 1, but that just kills the channel until you do a zap
restart. That being said, this is an option
exten = _911,1,System('/usr/sbin/asterisk -rx zap restart')
exten = _911,2,System('/usr/sbin/asterisk -rx zap restart')
Second instance is to start the line that was in use during first restart
exten = _911,3,Dial(Zap/1/ww911,60)
exten = _911,3,Hangup()
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov
Sent: Friday, November 28, 2008 4:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] force channel hangup
Because hangup (and other behavioural) directives can only be addressed
to a particular instance of a channel use, i.e.
Technology/channel-uniqueID. The latter are not addressable from the
dial plan except implicitly.
Danny Nicholas wrote:
Why wouldn't this work?
exten = _911,1,Hangup(Zap/1)
exten = _911,2,Dial(Zap/1/ww911,60)
exten = _911,3,Hangup()
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony
Messina
Sent: Friday, November 28, 2008 3:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] force channel hangup
On Thursday 27 November 2008 20:25:49 Kelvin Chan wrote:
Hi guys,
I have 1 zap channel in my house shared among couple people. If someone
dials 911, I want that zap channel to be disconnected right away to make
way for the 911 call.
I dug through voip-info.org and didn't find much.
Any hints?
i use this:
http://messinet.com/index.php?page_name=Asteriskwikipage=Asteriske911
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
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Re: [asterisk-users] force channel hangup
On Fri, Nov 28, 2008 at 04:42:01PM -0600, Danny Nicholas wrote:
Right you are, Alex. How about (CLI) Zap restart? I was thinking zap
destroy channel 1, but that just kills the channel until you do a zap
restart. That being said, this is an option
exten = _911,1,System('/usr/sbin/asterisk -rx zap restart')
exten = _911,2,System('/usr/sbin/asterisk -rx zap restart')
This will disconnect all existing Zap calls.
BTW: As of Asterisk 1.4.22 / 1.6.0 'dahdi restart' actually works as
promised and you don't need to run it twice.
Second instance is to start the line that was in use during first restart
exten = _911,3,Dial(Zap/1/ww911,60)
exten = _911,3,Hangup()
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] force channel hangup
On Fri, Nov 28, 2008 at 05:24:36PM -0500, Alex Balashov wrote: Because hangup (and other behavioural) directives can only be addressed to a particular instance of a channel use, i.e. Technology/channel-uniqueID. The latter are not addressable from the dial plan except implicitly. For a Zap channel the unique ID will mostly be '1' . In some cases it will be '2'. So: exten = _911,1,Hangup(Zap/1-1) exten = _911,n,Hangup(Zap/1-2) exten = _911,n,Dial(Zap/1/ww911,60) exten = _911,n,Hangup() I wonder, though, how long does it take for the hangup to take effect. A hangup requests the channel to hang up. This is done later in the channel context. I wonder if it is normally done quickly enough. -- 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
[asterisk-users] received wrong state events for originate command
Hey all, Something is wrong when i use originate command to call my phone (Asterisk1.4.22 + xp100 card). Actually, i have two problems. The first one: If i fire a outgoing call using originate command directly, after my pc startup, i will receive below error message: [Nov 26 07:58:53] NOTICE[6559]: channel.c:2898 __ast_request_and_dial: Unable to request channel Zap/1/13x but i can call the FXO using my phone, everything seems perfect! After the incomming call, i fire outgoing call using originate again, it works now, my phone can ring, i also can pick up it(I seems originate did not create a new Zap channel,just used an exsiting channel?). But the second problem produced, i received the Dialing, UP, Newexten events before my phone ringing. It is supposed that i send an originate command (like Dial application), the last state should be Dialing... until i pick up my phone or timeout. These problems only for Zap channel, if i fire a outgoing call to SIP channel, it works well. What wrong with me ? Here is my php script: $socket = fsockopen(127.0.0.1,5038,$errno,$errstr,$timeout); fputs($socket,Action: Login\r\n); fputs($socket,Username: tester\r\n); fputs($socket,Secret: test\r\n\r\n); fputs($socket,Action: Originate\r\n); fputs($socket,Channel: Zap/1/13\r\n); fputs($socket,Context: callme\r\n); fputs($socket,Exten: s\r\n); fputs($socket,Priority: 1\r\n\r\n); fclose($socket); Best regards, Xiaoshuang ___ -- 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] Trixbox 2.6.1.13 OpenR2
*Good morning! * *I verified that the trixbox version Trixbox 2.6.1.13 has support for OpenR2, I verified in the repository that has to libraries of the project openR2, but I don't manage to do to work in the trixbox, when I type the command (it colors show channeltypes)ele no demostra the support to MFC+R2, they could help finding out which package is necessary of the trixbox and which the necessary configurations that should make! I have been installing the trixbox version 2.6.1.13 and a Digium 110p, they put in the trixbox only get to do to work in ISDN! Thank you very much* ___ -- 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] Trixbox 2.6.1.13 OpenR2
Good morning! I verified that the trixbox version Trixbox 2.6.1.13 has support for OpenR2, I verified in the repository that has to libraries of the project openR2, but I don't manage to do to work in the trixbox, when I type the command (show channeltypes) he doesn't demonstrate the support to MFC+R2, they could help finding out which package is necessary of the trixbox and which the necessary configurations that should make! I have been installing the trixbox version 2.6.1.13 and a Digium 110p, they put in the trixbox only get to do to work in ISDN! Thank you very much ___ -- 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] libspandsp.so.0: cannot open shared object file: No such file or directory
libspandsp.so.0: cannot open shared object file: No such file or directory Created the symlink: /usr/local/lib# ls -lt lib* lrwxrwxrwx 1 root staff 19 2008-11-28 22:42 libspandsp.so.0 - libspandsp.so.1.0.0 -rw-r--r-- 1 root staff 1849266 2008-11-13 13:26 libspandsp.a -rwxr-xr-x 1 root staff 865 2008-11-13 13:26 libspandsp.la lrwxrwxrwx 1 root staff 19 2008-11-13 13:26 libspandsp.so - libspandsp.so.1.0.0 lrwxrwxrwx 1 root staff 19 2008-11-13 13:26 libspandsp.so.1 - libspandsp.so.1.0.0 -rwxr-xr-x 1 root staff 1433877 2008-11-13 13:26 libspandsp.so.1.0.0 Edited /etc/ld.so.conf: # Begin -- /etc/ld.so.conf include /etc/ld.so.conf.d/*.conf /usr/local/lib # End: --- /etc/ld.so.conf Googled the heck out of it: http://www.google.com/search?q=libspandsp.so.0:+cannot+open+shared+object Still can't find the answer. Any ideas? ___ -- 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] Trixbox 2.6.1.13 OpenR2
On Sat, Nov 29, 2008 at 10:18 AM, Yuri [EMAIL PROTECTED] wrote: *Good morning! * *I verified that the trixbox version Trixbox 2.6.1.13 has support for OpenR2, I verified in the repository that has to libraries of the project openR2, but I don't manage to do to work in the trixbox, when I type the command (it colors show channeltypes)ele no demostra the support to MFC+R2, they could help finding out which package is necessary of the trixbox and which the necessary configurations that should make! I have been installing the trixbox version 2.6.1.13 and a Digium 110p, they put in the trixbox only get to do to work in ISDN! Thank you very much* Hi Yuri, I also read that 2.6.1.13 would have OpenR2 support built in but found that this was not entirely true. The library package is in the repository, but support for OpenR2 is not in the provided Asterisk package. I ended up downloading the source and recompiling from the OpenR2 site. //Peter ___ -- 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] libspandsp.so.0: cannot open shared object file: No such file or directory
Paste 'ldd /usr/sbin/asterisk'. Doug wrote: libspandsp.so.0: cannot open shared object file: No such file or directory Created the symlink: /usr/local/lib# ls -lt lib* lrwxrwxrwx 1 root staff 19 2008-11-28 22:42 libspandsp.so.0 - libspandsp.so.1.0.0 -rw-r--r-- 1 root staff 1849266 2008-11-13 13:26 libspandsp.a -rwxr-xr-x 1 root staff 865 2008-11-13 13:26 libspandsp.la lrwxrwxrwx 1 root staff 19 2008-11-13 13:26 libspandsp.so - libspandsp.so.1.0.0 lrwxrwxrwx 1 root staff 19 2008-11-13 13:26 libspandsp.so.1 - libspandsp.so.1.0.0 -rwxr-xr-x 1 root staff 1433877 2008-11-13 13:26 libspandsp.so.1.0.0 Edited /etc/ld.so.conf: # Begin -- /etc/ld.so.conf include /etc/ld.so.conf.d/*.conf /usr/local/lib # End: --- /etc/ld.so.conf Googled the heck out of it: http://www.google.com/search?q=libspandsp.so.0:+cannot+open+shared+object Still can't find the answer. Any ideas? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] Anonymous callerid
Hi Thanks for your reply. Actully we are getting the anonymous callerid from the originated phone (blocked from phone) so we need to override the callerid and then pass to network. we need to send out caller id. That is why we tried to override it. But we are not able to override it. Please help for this! Thanks, Max Alex Voip Developer On Fri, Nov 28, 2008 at 7:47 PM, Philipp Kempgen [EMAIL PROTECTED]wrote: Max Alex schrieb: I have one issue regarding override callerid when i have anonymous call. I have added PAI in sip header and also set sendrpid = yes in sip.conf but the callerid is not overriding while i am sending call to three digit calling like 911. The caller ID sent to emergency or law enforcement numbers is network-provided not user-provided so you can't override it. Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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
