Re: [Asterisk-Users] Phones were working fine - Now there is no audiowhen calling between extensions
On Mar 20, 2006, at 4:39 PM, Gabriel Afana wrote: I just did a little RTP debug and this is what it shows: == Spawn extension (304, 301, 1) exited non-zero on 'SIP/304-d9f8' -- Accepting AUTHENTICATED call from 216.152.244.81: requested format = ulaw, requested prefs = (), actual format = ulaw, host prefs = (), priority = mine -- Executing Dial(IAX2/to_80-1, SIP/301) in new stack -- Called 301 -- SIP/301-1fec is ringing -- SIP/301-1fec answered IAX2/to_80-1 Got RTP packet from 24.50.66.128:2228 (type 0, seq 16810, ts 344311448, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16811, ts 344311608, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16812, ts 344311768, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16813, ts 344311928, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16814, ts 344312088, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16815, ts 344312248, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16816, ts 344312408, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16817, ts 344312568, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16818, ts 344312728, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16819, ts 344312888, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16820, ts 344313048, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16821, ts 344313208, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16822, ts 344313368, len 160) .. that goes on for ever while the call is in progress. This is a call between phones that go between two * servers. If I make a call between phones both registered to the same asterisk server, this is my RTP stream: -- Executing Dial(SIP/304-c211, SIP/301|30|r) in new stack -- Called 301 -- SIP/301-b2c8 is ringing -- SIP/301-b2c8 answered SIP/304-c211 -- Attempting native bridge of SIP/304-c211 and SIP/301-b2c8 Got RTP packet from 24.50.66.128:2234 (type 0, seq 39729, ts -1972065425, len 160) Sent RTP packet to 24.50.66.128:2232 (type 0, seq 24819, ts 0, len 160) Got RTP packet from 24.50.66.128:2234 (type 0, seq 39730, ts -1972065265, len 160) Sent RTP packet to 24.50.66.128:2232 (type 0, seq 24820, ts 160, len 160) Got RTP packet from 24.50.66.128:2234 (type 0, seq 39731, ts -1972065105, len 160) Sent RTP packet to 24.50.66.128:2232 (type 0, seq 24821, ts 320, len 160) Got RTP packet from 24.50.66.128:2234 (type 0, seq 39732, ts -1972064945, len 160) Sent RTP packet to 24.50.66.128:2232 (type 0, seq 24822, ts 480, len 160) Got RTP packet from 24.50.66.128:2232 (type 0, seq 46694, ts 1105329892, len 160) Sent RTP packet to 24.50.66.128:2234 (type 0, seq 42690, ts 0, len 160) Got RTP packet from 24.50.66.128:2232 (type 0, seq 46695, ts 1105330052, len 160) Sent RTP packet to 24.50.66.128:2234 (type 0, seq 42691, ts 160, len 160) Got RTP packet from 24.50.66.128:2232 (type 0, seq 46696, ts 1105330212, len 160) Sent RTP packet to 24.50.66.128:2234 (type 0, seq 42692, ts 320, len 160) Got RTP packet from 24.50.66.128:2232 (type 0, seq 46697, ts 1105330372, len 160) Sent RTP packet to 24.50.66.128:2234 (type 0, seq 42693, ts 480, len 160) Got RTP packet from 24.50.66.128:2232 (type 0, seq 46698, ts 1105330532, len 160) Sent RTP packet to 24.50.66.128:2234 (type 0, seq 42694, ts 640, len 160) [THE END] Once I anser the call, the RTP string starts and then stops right where I put [THE END]. Did you try setting reinvite to no? Seems the native bridge is what's failing. Rethink your routing with regards native bridging (ie everybody is able to get through there nats and be identified? I don't really know, I am only trying to be helpful. Hope it's worth something. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Phones were working fine - Now there is noaudiowhen calling between extensions
Thanks for the response. Yes, canreinvite is set to no on all lines. After some testing, I was able to get sound between phones when they were both registered to the same server. Maybe the IAX trunk is messing something up. strange because it was working perfect last week and nothing changed! - Gabe Did you try setting reinvite to no? Seems the native bridge is what's failing. Rethink your routing with regards native bridging (ie everybody is able to get through there nats and be identified? I don't really know, I am only trying to be helpful. Hope it's worth something. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] answer delay
Hi,I've tryed it using my mobile and I've been charged.Maybe, my mobile operator(Vodafone) does not support it?Thanks again.p.s.: hi John, I love to learn(books, google, lists, ecc...) and cooperation and I can say that everytime I learn something new. :o) 2006/3/21, CC Jay [EMAIL PROTECTED]: Not sure about 1.2.4 but with 1.0.9, you'll need to add noanswer to playback since playback will try to answer the line, i.e.,exten = 5551234,1,Playback(you-wont-be-billed-for-hearing-this, noanswer) exten = 5551234,n,Answeretc. ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- .:FaberK:. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream unit HT-488
hi, if interested please consider the TigerNetcom box of 104 for doing the same functionality, much better piece and at considerable lower price. for technical information on how to use it i would be happy to assist off list. On 3/20/06, Martin Joseph [EMAIL PROTECTED] wrote: On Mar 19, 2006, at 2:04 PM, Oliver Vermeulen wrote: Hi All, Anybody knows how to terminated calls using Grandstream Ht488 and the FXO port ? I can ring the FXO port fine , rings 1once then give me dial tone. I had:exten = _NXX,1,Dial(SIP/@2003,60,D(w$EXTEN}))exten = _NXX,2,HangupWhere 2003 was the extension of the FXO on the HT-488.This worked ok for dialing 7 digit calls to the FXO, but also had a weird double (oneafter the other) ringback?Also use dtmfmode=RFC2833 in the extension and set the HT-488 the same.I had to give up on that device due to poor audio quality and echo issues .Also intermittent hanging made this device unacceptable forme.Let us know if it works for you?Also which firmware and asteriskversion are you using?Marty___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Do Not Disturb?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... You can do the same thing with DND. Turn the value on or off, then in your dial string, check the database value and act accordingly. Hi Doug. Do you know how to, when leaving office, set all incoming calls to transfer do my coworker? -- Tomislav Parcina tparcina#lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CDR problem with TAPI
Hi, we just noticed a strange CDR problem. We are using individual phone numbers for all our SIP phones. During dialout we do a database lookup in order to set the correct callerid (e.g. phone has number 100 but in external calls this should be displayed as CID -20). This works like a charm and the CDRs look correct. When we start a call using TAPI (e.g. AstTapi) the call setup is a bit different: I start the call, my phone (100) rings, I fetch the call and Asterisk dials the intended number. In the LOG I can see that the callerID is again set correctly to 20. But the CDR now does not show from 20 to 1234567 but from 100 to 1234567 ignoring the Set(CALLERID(number)=20) completly. Bug? Feature? Kind regards, JP smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queue and busy/congested ZAP channels
Hi, I'm having a problem with the queue behaviour in my place: I have two ISDN channels to the outside (Zap/1) and two channels two a Siemens Gigaset (Zap/4). I also use a SIP gateway to call outside and have a couple of IP phones around as well (SIP). The Gigaset has about 5 phones connected to it (+base station). Whenever two people are using those, I always am blocking two internal channels, so users who call in from the outside always get a busy or the voicemail. I configured a queue to take all calls to our central number, so people will get to a phone as soon as possible. However, when there are only agents logged on from the Zap/4 line, and both channels are used, the caller gets transferred to the voicemail of one of the agents instead of beeing put into the queue. Here is some console output on those events: -- Accepting voice call from '1797808366' to '12298890' on channel 0/1, span 1 -- Executing Queue(Zap/1-1, reception) in new stack -- Started music on hold, class 'default', on channel 'Zap/1-1' -- outgoing agentcall, to agent '1003', on 'Local/[EMAIL PROTECTED],1' -- Called Agent/1003 -- outgoing agentcall, to agent '1000', on 'Local/[EMAIL PROTECTED],1' -- Called Agent/1000 -- Executing Macro(Local/[EMAIL PROTECTED],2, call-user| 1003) in new stack -- Executing Dial(Local/[EMAIL PROTECTED],2, Zap/g4/1003Sip/S1003Sip/1003|45|tr) in new stack Mar 21 09:57:38 NOTICE[4354]: app_dial.c:1030 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circu it/channel congestion) -- Executing Macro(Local/[EMAIL PROTECTED],2, call-user| 1000) in new stack -- Executing Dial(Local/[EMAIL PROTECTED],2, Zap/g4/1000Sip/S1000Sip/1000|45|tr) in new stack Mar 21 09:57:38 NOTICE[4356]: app_dial.c:1030 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circu it/channel congestion) Mar 21 09:57:38 WARNING[4354]: chan_sip.c:1973 create_addr: No such host: S1003 Mar 21 09:57:38 NOTICE[4354]: app_dial.c:1030 dial_exec_full: Unable to create channel of type 'Sip' (cause 3 - No rou te to destination) Mar 21 09:57:38 WARNING[4356]: chan_sip.c:1973 create_addr: No such host: S1000 Mar 21 09:57:38 NOTICE[4356]: app_dial.c:1030 dial_exec_full: Unable to create channel of type 'Sip' (cause 3 - No rou te to destination) Mar 21 09:57:38 WARNING[4354]: chan_sip.c:1973 create_addr: No such host: 1003 Mar 21 09:57:38 NOTICE[4354]: app_dial.c:1030 dial_exec_full: Unable to create channel of type 'Sip' (cause 3 - No rou te to destination) == Everyone is busy/congested at this time (3:0/1/2) -- Executing VoiceMail(Local/[EMAIL PROTECTED],2, u1003) in new stack -- Playing 'vm-theperson' (language 'en') Any ideas why it goes to voicemail instead of keeping the caller in the queue? Thanks, Christian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE : RE : [asterisk-dev] iax failure?
[EMAIL PROTECTED] a écrit : Oops ! I have upgraded TRUNK again via SVN and all was seeming to be fine, no more invalid IAX2 frames and able to place and receive calls. I was happy.. But, few calls later (about 5 minutes) : INVAL frames again and no more possibility to place or receive calls, no prompt tone, nothing ! Strange... Best Regards, Francois BERGERET, France. Hi Francois, look at this http://bugs.digium.com/view.php?id=6748 perhaps also your case -- Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to make extension groups ???
Hello All,i am repeating this question for the sixth time but i think i was not explaining the problem correctly. . Now i will try to explain it..I have 4 telephone lines(PSTN) in my PBX.Now I want to makegroups of the extensions to use that lines. e.g. extensions 12,13 31 are in groupA extensions 14 - 20 are in groupB extensions 21 - 30 are in groupCgroupA has access on lines 2,3,4 (Try line 2, if busy try line 3 ,if busy try line 4) groupB has access on lines 3,4 (Try line 3 ,if busy try line 4) groupC has access on line 4 only. (Try line 4 only, and if busy give busy tone)Line 1 is reserved for one extension only. i.e. 11 I will be grateful for an early and complete response.Thanks a lotFaisal Yahoo! Mail Use Photomail to share photos without annoying attachments.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to make groups of extensions ???
Hello All,i am repeating this question for the sixth time but i think i was not explaining the problem correctly. . Now i will try to explain it..I have 4 telephone lines(PSTN) in my PBX.Now I want to makegroups of the extensions to use that lines. e.g. extensions 12,13 31 are in groupA extensions 14 - 20 are in groupB extensions 21 - 30 are in groupCgroupA has access on lines 2,3,4 (Try line 2, if busy try line 3 ,if busy try line 4) groupB has access on lines 3,4 (Try line 3 ,if busy try line 4) groupC has access on line 4 only. (Try line 4 only, and if busy give busy tone)Line 1 is reserved for one extension only. i.e. 11 I will be grateful for an early and complete response.Thanks a lotFaisal Yahoo! Mail Use Photomail to share photos without annoying attachments.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to make groups of extensions ???
RTFM El Martes, 21 de Marzo de 2006 10:53, Faisal Inam escribió: Hello All, i am repeating this question for the sixth time but i think i was not explaining the problem correctly. . Now i will try to explain it.. I have 4 telephone lines(PSTN) in my PBX. Now I want to make groups of the extensions to use that lines. e.g. extensions 12,13 31 are in groupA extensions 14 - 20 are in groupB extensions 21 - 30 are in groupC groupA has access on lines 2,3,4 (Try line 2, if busy try line 3 ,if busy try line 4) groupB has access on lines 3,4 (Try line 3 ,if busy try line 4) groupC has access on line 4 only. (Try line 4 only, and if busy give busy tone) Line 1 is reserved for one extension only. i.e. 11 I will be grateful for an early and complete response. Thanks a lot Faisal - Yahoo! Mail Use Photomail to share photos without annoying attachments. -- Francisco J. Pérez Botella ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Aterisk with Realtime
Hi i have installed unixODBC Drivers and created DSN but when i reload from * console asterisk terminating the services if i do again start the services and change something and reload again its hangs is that bug in 1.2.5 ram On 3/21/06, Aaron Daniel [EMAIL PROTECTED] wrote: Since you say you're using mysql as the backend, you need to changeanything that says odbc to mysql so that the server knows where to find the db at.Also, you need to make sure the DB info is inres_mysql.conf.Aaronram wrote: Hi thanks for the reply this what my extconfig sipusers = odbc,asterisk,2_Sip sippeers = odbc,asterisk,2_Sip extensions = odbc,asterisk,2_Extensions voicemail= odbc,asterisk,2_VMUsers voicemail_messages = odbc,asterisk,2_VM waht is wrong with this ? ram On 3/20/06, *Aaron Daniel* [EMAIL PROTECTED] mailto: [EMAIL PROTECTED] wrote: That, and make sure you've got extconfig set to use mysql for it's sippusers and sippeers and not odbc. Aaron Patrick wrote: On Mon, 2006-03-20 at 23:14 +0530, ram wrote: Hi iam working with asterisk with mysql Realtime when i have confgured and run the asterisk iam getting the following error i dig all the places for help could not find the results could some one help me what is wrong iam using 1.2.5 on FC4 Mar 20 23:04:52 NOTICE[2054] cdr.c: CDR simple logging enabled. Mar 20 23:04:52 NOTICE[2054] indications.c: Removed default indication country 'us' Mar 20 23:04:52 WARNING[2054] pbx_config.c: The use of '_.' for an extension is strongly discouraged and can have unexpected b ehavior.Please use '_X.' instead at line 3 Read the message and do as it suggests: in your dialplan replace all _. with _X. Patrick ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Do Not Disturb?
Tomislav Parčina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... You can do the same thing with DND. Turn the value on or off, then in your dial string, check the database value and act accordingly. Hi Doug. Do you know how to, when leaving office, set all incoming calls to transfer do my coworker? The phones that I've worked with have this function built-in. Cisco, Polycom, etc. You should be able to do forwarding on the phone. If not, you should be able to find the necessary info on the Wiki. http://www.voip-info.org/wiki-Asterisk+-+documentation+of+application+commands Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] stop monitor on transfer
Anton Krall wrote: Guys. This idea has been banging my headfor days now and I feel the need to share with you. Imagine this scenario: all calls come in thru a receptionist, asterisk records all incoming calls, the receptionist's work is to transfer the calls to internal people but some of them are bosses and you know how bosses are, they don't want their calls to be recorded, so, I have been trying to figure a way on how to stop monitoring / recoring calls once they are transferred to a bosses extension while othe transferd to other people stay on record mode. Anton, hi; I've got exactly the opposite problem. I *want* to record the call after the transfer, but (using MixMonitor and SIP transfers on Snom handsets) the recording terminates with the transfer. Are you using Asterisk native transfer ? jd ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with chan_iax.c implimentation causes bad audio?
Matt wrote: I received an e-mail from a vendor who says: We have recently become aware of an issue in the chan_iax2 implementation of IAX2. This issue leads to degraded audio quality. Due to this we are urging everyone to move to SIP. I don't want to discount what this person is talling me, but I'm curious to know why I would only be having issues connecting to his servers, and also what exactly the issue is (if anyone knows). I was always under the impression that IAX2 was a better way to connect servers and was more advanced (jitterbuffer/etc) then sip was. Can anyone comment on this? There have been a number of interoperability issues with iax over the last year or so. It seems the majority are related to bugs associated with counter rollovers, jitterbuffer changes, frames sent with identical counters/timestamps, dtmf encoding, issues with certain codecs, etc. I'd hate to have the job of creating a matrix of which * versions function with other versions knowing full well that multiple changes occurred between versions. If you search the bug tracker for open closed iax issues, you'll see a number of them. (Note: not all iax changes came through the bug tracker either.) Add to that the fact that iax is actually a proprietary protocol implementation (eg, not based on any current published/approved standards), and the fact that only folks that run asterisk actually use the protocol, you now have a fairly major support issue from the itsp's perspective. Couple all of the above with how many newbies try to implement an * system with almost zero knowledge of how to implement or support their own system, and its not difficult to understand why the itsp's have a support issue with iax. Given the majority of itsp's have had to modify source code to address their own operational/business objectives, its not at all easy for them to keep up to date with asterisk releases patches. Compare that to the stability of the underlying sip/rtp protocols and I think you'll reach a conclusion that is similar to the itsp that told you that. FWIW, I'll continue to use iax with my itsp's. ;) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with chan_iax.c implimentation causes bad audio?
All of what you say is true, but wouldn't one expect a business who has wrapped themselves with Asterisk would be better able to provide IAX ? One wonders about their long term viability, given this position and the condition of their website. Broken links, and such. JMO John Novack Rich Adamson wrote: Matt wrote: I received an e-mail from a vendor who says: We have recently become aware of an issue in the chan_iax2 implementation of IAX2. This issue leads to degraded audio quality. Due to this we are urging everyone to move to SIP. I don't want to discount what this person is talling me, but I'm curious to know why I would only be having issues connecting to his servers, and also what exactly the issue is (if anyone knows). I was always under the impression that IAX2 was a better way to connect servers and was more advanced (jitterbuffer/etc) then sip was. Can anyone comment on this? There have been a number of interoperability issues with iax over the last year or so. It seems the majority are related to bugs associated with counter rollovers, jitterbuffer changes, frames sent with identical counters/timestamps, dtmf encoding, issues with certain codecs, etc. I'd hate to have the job of creating a matrix of which * versions function with other versions knowing full well that multiple changes occurred between versions. If you search the bug tracker for open closed iax issues, you'll see a number of them. (Note: not all iax changes came through the bug tracker either.) Add to that the fact that iax is actually a proprietary protocol implementation (eg, not based on any current published/approved standards), and the fact that only folks that run asterisk actually use the protocol, you now have a fairly major support issue from the itsp's perspective. Couple all of the above with how many newbies try to implement an * system with almost zero knowledge of how to implement or support their own system, and its not difficult to understand why the itsp's have a support issue with iax. Given the majority of itsp's have had to modify source code to address their own operational/business objectives, its not at all easy for them to keep up to date with asterisk releases patches. Compare that to the stability of the underlying sip/rtp protocols and I think you'll reach a conclusion that is similar to the itsp that told you that. FWIW, I'll continue to use iax with my itsp's. ;) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Caller ID forwarding with Pickup() application?
Hi,
I'm using the Pickup() application for direct call pickup having the following
line in the dialplan:
exten = _*88XX,1,Pickup(${EXTEN:2})
It works OK, though I would like to have to get the original caller ID number
forwarded to the phone where I do the pickup and have it displayed during the
call. Currently the string *88xx remains on the screen of the phone I do the
pickup. It is a snom 320. I'm using asterisk 1.2.4.
Any ideas?
Thanks,
-Tamas
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[Asterisk-Users] app_queue and ARA
Hello, I've configured ACD with ARA asterisk-1.2.4 . I try show queues command but no queue is shown. why ? Can I keep the caller on queue until an agent answer the call ? I use ARA to configure queues and members however i have to use agents.conf to store the agents. I wish to configure agents in SQL db. Is it possible ? Regards Harry ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Web-ex type solution for use with asterisk
Is there an app or softphone for meetings that displays the hosts screen like webex or intercall. Jordan Novak ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with chan_iax.c implimentation causes bad audio?
On Tuesday 21 March 2006 07:19, Matt wrote: I was going to avoid naming names :P But anyway.. yes it's asterlink. Guys seem nice enough.. and by golly.. when I switched to SIP the termination is crystal clear... so far I'm happy with the service from Asterlink... just wish I could use IAX2 oh well.. it really matters not to me HOW I get the audio stream.. just that it works and is stable. I don't know why you'd avoid naming names. Asterlink does have good service, and as I said they are a smart bunch of guys. I get troubles with my SIP registrations to them on occasion but that's it. I have absolutely no trouble recommending them to anyone. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with chan_iax.c implimentation causes bad audio?
I don't know why you'd avoid naming names. Asterlink does have good service, and as I said they are a smart bunch of guys. I get troubles with my SIP registrations to them on occasion but that's it. I have absolutely no trouble recommending them to anyone. Hi, Wanted to avoid naming names to keep the peace :) Quick question. Do you have any 'delay' with asterlink? Audio FROM ME to THEM is almost instant. But SIP audio from THEM to ME has about a 2 second delay in it. Any thoughts on that? Brian and I are currently trying to trouble shoot it. Just wondered if you had the same problem? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Web-ex type solution for use with asterisk
On 3/21/06, Jordan Novak [EMAIL PROTECTED] wrote: Is there an app or softphone for meetings that displays the hosts screen like webex or intercall. http://www.btwtech.com/wipast/ -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice mail not working with Asteriks 1.2.5
Mazhar Hussain wrote:
Hi,
I have upgraded my PBX to Asterisk 1.2.5 , previously I was
using Asterisk 1.0.9, and Every thing was working fine ,But now
voice mail is not working. The error I am receiving in log files is
like following,
WARNING[2413] app_voicemail.c: No entry in voicemail config file for '12'
You now need to specify the context (I use sip) when leaving unavailable
or busy messages.
For example:
s,7,Voicemail(u${ARG1})
becomes
s,7,Voicemail([EMAIL PROTECTED])
Doug
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RE: [Asterisk-Users] Problem with chan_iax.c implimentation causesbad audio?
We have three remote call center Asterisk servers communicating with two central Asterisk boxes over a private IP-VPN with QoS. All systems were running Asterisk 1.0.7 communicating via IAX2 with little or no quality issues at all. Once we upgraded to Asterisk 1.2.4 call quality with IAX2 was horrific. We tried with/without jitterbuffer. We messed with every jitterbuffer parameter. We tried G729/ilbc/ulaw. It was a total mess. We switched to SIP and instantly all problems disappeared. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Tuesday, March 21, 2006 9:27 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Problem with chan_iax.c implimentation causesbad audio? On Tuesday 21 March 2006 07:19, Matt wrote: I was going to avoid naming names :P But anyway.. yes it's asterlink. Guys seem nice enough.. and by golly.. when I switched to SIP the termination is crystal clear... so far I'm happy with the service from Asterlink... just wish I could use IAX2 oh well.. it really matters not to me HOW I get the audio stream.. just that it works and is stable. I don't know why you'd avoid naming names. Asterlink does have good service, and as I said they are a smart bunch of guys. I get troubles with my SIP registrations to them on occasion but that's it. I have absolutely no trouble recommending them to anyone. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Web-ex type solution for use with asterisk
I happened to see a demonstration of WebInterpoint for Asterisk at the Digium booth at the recent VON show, and was impressed with the capabilities. Cory Andrews ++ VOIPSupply.com A Division of b2 Technologies 454 Sonwil Drive Buffalo, NY 14225 direct - 716.250.3402 mobile - 716.907.4059 email - [EMAIL PROTECTED] AIM - b2Cory - Original Message - From: BJ Weschke [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, March 21, 2006 9:43 AM Subject: Re: [Asterisk-Users] Web-ex type solution for use with asterisk On 3/21/06, Jordan Novak [EMAIL PROTECTED] wrote: Is there an app or softphone for meetings that displays the hosts screen like webex or intercall. http://www.btwtech.com/wipast/ -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voice mail not working with Asteriks 1.2.5
Hi, I have upgraded my PBX to Asterisk 1.2.5 , previously I was using Asterisk 1.0.9, and Every thing was working fine ,But now voice mail is not working. The error I am receiving in log files is like following, WARNING[2413] app_voicemail.c: No entry in voicemail config file for '12' I have searched for solution a lot can Any one of you let me know how can I solve this issue do I need to apply any patch for asterisk Here is voicemail.conf config file [general] format=wav49|gsm|wav serveremail=asterisk attach=yes skipms=3000 maxsilence=10 silencethreshold=128 maxlogins=3 saycid=yes [zonemessages] eastern=America/New_York|'vm-received' Q 'digits/at' IMp central=America/Chicago|'vm-received' Q 'digits/at' IMp central24=America/Chicago|'vm-received' q 'digits/at' H 'digits/hundred' M 'hours' [headoffice] 901=111, Arshed User, [EMAIL PROTECTED] 12 = 235, Mazhar User, [EMAIL PROTECTED] 6412 = 235, Mazhar User, [EMAIL PROTECTED] [other] 1234 = 5678,Company2 User,[EMAIL PROTECTED] And here is modules.conf file [modules] autoload=yes noload = app_intercom.so noload = chan_modem.so noload = chan_modem_aopen.so noload = chan_modem_bestdata.so noload = chan_modem_i4l.so load = res_musiconhold.so noload = chan_alsa.so [global] A quick response in this regard will be highly appreciated Thanks, Mazhar ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice mail not working with Asteriks 1.2.5
Hi,
Check your context you need to specify voicemail as [EMAIL PROTECTED]
(context seems to have been more tightly enforced since version 1.2 came
out). Below is an example of one of the macro I use for extensions...
[macro-stdexten]
exten = s,1,Set(DYNAMIC_FEATURES=automon)
exten = s,2,Dial(${ARG2},20,Ttw)
exten = s,3,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail([EMAIL PROTECTED])
exten = s-NOANSWER,2,Playback(thank-you-for-callinggoodbye)
exten = s-NOANSWER,3,Hangup
exten = s-BUSY,1,Voicemail([EMAIL PROTECTED])
exten = s-BUSY,2,Playback(thank-you-for-callinggoodbye)
exten = s-BUSY,3,Hangup
exten = s-CHANUNAVAIL,1,Voicemail([EMAIL PROTECTED])
exten = s-CHANUNAVAIL,2,Playback(thank-you-for-callinggoodbye)
exten = s-CHANUNAVAIL,3,Hangup
exten = _s-.,1,Goto(s-NOANSWER,1)
Hope this helps
Thanks
Mazhar Hussain wrote:
Hi,
I have upgraded my PBX to Asterisk 1.2.5 , previously I was
using Asterisk 1.0.9, and Every thing was working fine ,But now
voice mail is not working. The error I am receiving in log files is
like following,
WARNING[2413] app_voicemail.c: No entry in voicemail config file for '12'
I have searched for solution a lot can Any one of you let me know how
can I solve this issue do I need to apply any patch for asterisk
Here is voicemail.conf config file
[general]
format=wav49|gsm|wav
serveremail=asterisk
attach=yes
skipms=3000
maxsilence=10
silencethreshold=128
maxlogins=3
saycid=yes
[zonemessages]
eastern=America/New_York|'vm-received' Q 'digits/at' IMp
central=America/Chicago|'vm-received' Q 'digits/at' IMp
central24=America/Chicago|'vm-received' q 'digits/at' H
'digits/hundred' M 'hours'
[headoffice]
901=111, Arshed User, [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
12 = 235, Mazhar User, [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
6412 = 235, Mazhar User, [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
[other]
1234 = 5678,Company2 User,[EMAIL PROTECTED]
And here is modules.conf file
[modules]
autoload=yes
noload = app_intercom.so
noload = chan_modem.so
noload = chan_modem_aopen.so
noload = chan_modem_bestdata.so
noload = chan_modem_i4l.so
load = res_musiconhold.so
noload = chan_alsa.so
[global]
A quick response in this regard will be highly appreciated
Thanks,
Mazhar
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[Asterisk-Users] Zap--IAX codec?
Hi, at my Asterisk box, I have a few of IAX2 phones (configured with alaw/ulaw/gsm codecs, in this order) and a PRI E1 line. In iax.conf I hav: disallow=all allow=alaw allow=ulaw allow=gsm During some incoming call, I read at console: -- Executing Dial(Zap/2-1, IAX2/215|20|TtwW) in new stack -- Called 215 -- Call accepted by 10.97.1.7 (format ulaw) -- Format for call is ulaw -- IAX2/215-33 is ringing -- IAX2/215-33 answered Zap/2-1 Why I have 'Format for call is ulaw'? I'd like to have alaw but keep ulaw to accomodate errors in various configurations (if any, not here!). -- Mimmus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with chan_iax.c implimentation causesbad audio?
On Tuesday 21 March 2006 09:47, Adam Robins wrote: We have three remote call center Asterisk servers communicating with two central Asterisk boxes over a private IP-VPN with QoS. All systems were running Asterisk 1.0.7 communicating via IAX2 with little or no quality issues at all. Once we upgraded to Asterisk 1.2.4 call quality with IAX2 was horrific. We tried with/without jitterbuffer. We messed with every jitterbuffer parameter. We tried G729/ilbc/ulaw. It was a total mess. Did you upgrade all three boxes? Did you try disabling trunking? What was your last mile solution? (i.e. what did the end-users speak into, and how did their calls get to the PSTN?) If it was to a far-end Asterisk box, what version where they running? Were you communicating using IAX2 to them too? Did they upgrade to 1.2.4 as well? I am running SVN trunk with IAX2 and SIP and have *zero* issues. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller ID forwarding with Pickup() application?
Check this out:
http://lists.digium.com/pipermail/asterisk-users/2006-March/143394.html
When that one will work, then yours will.
On 3/21/06, Tamás Bondár [EMAIL PROTECTED] wrote:
Hi,
I'm using the Pickup() application for direct call pickup having the following
line in the dialplan:
exten = _*88XX,1,Pickup(${EXTEN:2})
It works OK, though I would like to have to get the original caller ID number
forwarded to the phone where I do the pickup and have it displayed during the
call. Currently the string *88xx remains on the screen of the phone I do the
pickup. It is a snom 320. I'm using asterisk 1.2.4.
Any ideas?
Thanks,
-Tamas
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Re: [Asterisk-Users] How to make groups of extensions ???
As others have told you already, RTFM. Context is what you are looking for. On 3/21/06, Faisal Inam [EMAIL PROTECTED] wrote: Hello All, i am repeating this question for the sixth time but i think i was not explaining the problem correctly. . Now i will try to explain it.. I have 4 telephone lines(PSTN) in my PBX. Now I want to make groups of the extensions to use that lines. e.g. extensions 12,13 31 are in groupA extensions 14 - 20 are in groupB extensions 21 - 30 are in groupC groupA has access on lines 2,3,4 (Try line 2, if busy try line 3 ,if busy try line 4) groupB has access on lines 3,4 (Try line 3 ,if busy try line 4) groupC has access on line 4 only. (Try line 4 only, and if busy give busy tone) Line 1 is reserved for one extension only. i.e. 11 I will be grateful for an early and complete response. Thanks a lot Faisal Yahoo! Mail Use Photomail to share photos without annoying attachments. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problem with chan_iax.c implimentation causesbadaudio?
We upgraded all five servers to 1.2.4. We tried trunking/notrunking. End users use an IAX2 softphone on their desktop PCs. Agents are VLANed and all IAX2 traffic is QoS'd on all LAN and WAN legs. Calls flow from the agents to the local Asterisk server as IAX2/ulaw. Then they went over the IP-VPN as IAX2/g729 (we tried ilbc and straight ulaw as well). Calls get to the PSTN from the central site via PRI on TE410P cards. Point is that it worked fine for 6-9 months before the Asterisk 1.2.4 upgrade. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Tuesday, March 21, 2006 10:21 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Problem with chan_iax.c implimentation causesbadaudio? On Tuesday 21 March 2006 09:47, Adam Robins wrote: We have three remote call center Asterisk servers communicating with two central Asterisk boxes over a private IP-VPN with QoS. All systems were running Asterisk 1.0.7 communicating via IAX2 with little or no quality issues at all. Once we upgraded to Asterisk 1.2.4 call quality with IAX2 was horrific. We tried with/without jitterbuffer. We messed with every jitterbuffer parameter. We tried G729/ilbc/ulaw. It was a total mess. Did you upgrade all three boxes? Did you try disabling trunking? What was your last mile solution? (i.e. what did the end-users speak into, and how did their calls get to the PSTN?) If it was to a far-end Asterisk box, what version where they running? Were you communicating using IAX2 to them too? Did they upgrade to 1.2.4 as well? I am running SVN trunk with IAX2 and SIP and have *zero* issues. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Phones were working fine - Now there is noaudiowhen calling between extensions
Polycoms are not the best if you want a phone that works behind NAT. On 3/21/06, Gabriel Afana [EMAIL PROTECTED] wrote: Thanks for the response. Yes, canreinvite is set to no on all lines. After some testing, I was able to get sound between phones when they were both registered to the same server. Maybe the IAX trunk is messing something up. strange because it was working perfect last week and nothing changed! - Gabe Did you try setting reinvite to no? Seems the native bridge is what's failing. Rethink your routing with regards native bridging (ie everybody is able to get through there nats and be identified? I don't really know, I am only trying to be helpful. Hope it's worth something. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with chan_iax.c implimentation causesbadaudio?
On Tuesday 21 March 2006 10:55, Adam Robins wrote: End users use an IAX2 softphone on their desktop PCs. Agents are VLANed If there were significant changes to chan_iax2 and these were not upgraded to match, this could explain the trouble. Point is that it worked fine for 6-9 months before the Asterisk 1.2.4 upgrade. Oh, I understand the point. I'm not defending a protocol change causing such breakage, I am just trying to identify why the breakage occurred when Asterisk was upgraded. Out of curiosity, which softphones do you use? What kind of interface to the user, just a cheap headset plugged into the speaker/mic on a soundcard (which soundcard? I've had trouble with some) or something fancier such as a Plantronics USB headset or bluetooth one? Regards, Andrew ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to make groups of extensions ???
On Tuesday 21 March 2006 10:53, C F wrote: As others have told you already, RTFM. Context is what you are looking for. Uh, I'm not exactly sure how contexts will help him here. Zaptel channel groups will help him. Contexts won't do shit here unless I'm grossly misinterpreting what he wants. Francisco's (and your) RTFM wasn't exactly helpful. Even in the Asterisk Handbook draft the concept of channel groups is *very* easy to miss. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Junghanns and Digium TDM400?
Hi all, is it possible to bridge a call between a Junghanns quadBRI card and a TDM400 in the same server? It should be I think, -- I am trying this and when an incoming call comes in, it hangs both up at the moment the bridge is attempted (and a subsequent 'qozap: dropped audio' error is show in the /var/log/messages) Any thoughts appreciated -- I've seen posts, but no clear results/solutions -- Chris Earle System Solutions Specialist -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problem with chan_iax.c implimentationcausesbadaudio?
All switches and routers give highest priority to traffic on IAX2 port 4569. We use DSCB values over the IP-VPN to prioritize it as well. This did not change with the upgrade, as we can still see proper packet coding. The softphone is provided by our vendor Aheeva. It is the same IAX2 softphone they use in their own call centers. Funny thing is that they say that moving to Asterisk 1.2.4 tremendously IMPROVED their call quality with IAX2. Headsets are Plantronics H251N tops with DA60 USB adapters. All Desktops are at least 2.0 GHz P4 with 512MB RAM -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Tuesday, March 21, 2006 11:08 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Problem with chan_iax.c implimentationcausesbadaudio? On Tuesday 21 March 2006 10:55, Adam Robins wrote: End users use an IAX2 softphone on their desktop PCs. Agents are VLANed If there were significant changes to chan_iax2 and these were not upgraded to match, this could explain the trouble. Point is that it worked fine for 6-9 months before the Asterisk 1.2.4 upgrade. Oh, I understand the point. I'm not defending a protocol change causing such breakage, I am just trying to identify why the breakage occurred when Asterisk was upgraded. Out of curiosity, which softphones do you use? What kind of interface to the user, just a cheap headset plugged into the speaker/mic on a soundcard (which soundcard? I've had trouble with some) or something fancier such as a Plantronics USB headset or bluetooth one? Regards, Andrew ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to make groups of extensions ???
On 3/21/06, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Tuesday 21 March 2006 10:53, C F wrote: As others have told you already, RTFM. Context is what you are looking for. Uh, I'm not exactly sure how contexts will help him here. Zaptel channel groups will help him. Contexts won't do shit here unless I'm grossly misinterpreting what he wants. Francisco's (and your) RTFM wasn't exactly helpful. Even in the Asterisk Handbook draft the concept of channel groups is *very* easy to miss. I disagree that it is very easy to miss, in fact even just mentioning it, makes it very easy to not miss, becuase it's a very understandable feature. How will groups without context help him? it's actualy both that he needs, however he will be able to get by without groups, but not without contexts. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk in the E1/R2 digital compelled enviroment
On Tue, 21 Mar 2006 12:18:20 +0800, Ganbaa wrote Hi all, I would like to use Asterisk in the E1/R2 digital compelled enviroment. Which card is better TE210P Dual T1/E1 card or Sangoma A102U Dual T1/E1? I heard Asterisk's Unicall add-on can support R2 signalling. But I dont have no idea. Would you give me advice? Unicall works well enough with Asterisk but you will find a couple things that are hard to get around. FIrst you must test the local variant of R2, right now Unicall supports around 10 to 15 variants. I hava only used Digium cards so I do not know if Sangoma makes a real difference. A friend of mine says they are better, but I do not have any evidence of that. To install Unicall just follow the instructions in http://www.soft-switch.org/unicall/installing-mfcr2.html and you should be able to compile Asterisk with all the extras. -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Aterisk with Realtime
Are you moving your db over to an odbc connection? Aaron On Tue, 21 Mar 2006, ram wrote: Hi i have installed unixODBC Drivers and created DSN but when i reload from * console asterisk terminating the services if i do again start the services and change something and reload again its hangs is that bug in 1.2.5 ram On 3/21/06, Aaron Daniel [EMAIL PROTECTED] wrote: Since you say you're using mysql as the backend, you need to change anything that says odbc to mysql so that the server knows where to find the db at. Also, you need to make sure the DB info is in res_mysql.conf. Aaron ram wrote: Hi thanks for the reply this what my extconfig sipusers = odbc,asterisk,2_Sip sippeers = odbc,asterisk,2_Sip extensions = odbc,asterisk,2_Extensions voicemail = odbc,asterisk,2_VMUsers voicemail_messages = odbc,asterisk,2_VM waht is wrong with this ? ram On 3/20/06, *Aaron Daniel* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: That, and make sure you've got extconfig set to use mysql for it's sippusers and sippeers and not odbc. Aaron Patrick wrote: On Mon, 2006-03-20 at 23:14 +0530, ram wrote: Hi iam working with asterisk with mysql Realtime when i have confgured and run the asterisk iam getting the following error i dig all the places for help could not find the results could some one help me what is wrong iam using 1.2.5 on FC4 Mar 20 23:04:52 NOTICE[2054] cdr.c: CDR simple logging enabled. Mar 20 23:04:52 NOTICE[2054] indications.c: Removed default indication country 'us' Mar 20 23:04:52 WARNING[2054] pbx_config.c: The use of '_.' for an extension is strongly discouraged and can have unexpected b ehavior. Please use '_X.' instead at line 3 Read the message and do as it suggests: in your dialplan replace all _. with _X. Patrick ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with chan_iax.c implimentationcausesbadaudio?
On Tuesday 21 March 2006 11:19, Adam Robins wrote: All switches and routers give highest priority to traffic on IAX2 port 4569. We use DSCB values over the IP-VPN to prioritize it as well. This did not change with the upgrade, as we can still see proper packet coding. Right, I wouldn't suspect otherwise. The softphone is provided by our vendor Aheeva. It is the same IAX2 softphone they use in their own call centers. Funny thing is that they say that moving to Asterisk 1.2.4 tremendously IMPROVED their call quality with IAX2. I wonder what the hell is going on then, that is definitely something strange. Headsets are Plantronics H251N tops with DA60 USB adapters. All Desktops are at least 2.0 GHz P4 with 512MB RAM Thanks for the information. I feel bad for not having a good solid answer for why it's occurring. As the saying goes: I don't have an answer, but I admire the problem... -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multiple processes
Title: Multiple processes Does anyone have any ideas why my recently updated * 1.2.5 system should spawn multiple * process at seemingly random intervals? Regards L:ee ###This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange.For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to make groups of extensions ???
On Tuesday 21 March 2006 11:28, C F wrote: I disagree that it is very easy to miss, in fact even just mentioning it, makes it very easy to not miss, becuase it's a very understandable feature. Well when you know what to look for everything is easy to find. :-) How will groups without context help him? it's actualy both that he needs, however he will be able to get by without groups, but not without contexts. *nothing* works without contexts, which is why I said the answer doesn't help. Contexts are for incoming calls, not outgoing ones. How do contexts help him? -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pickup problem
Ha -- this looks useful Just was trying to do a *8 on an IAXy phone...realized it didn't work across protocols If I implement this, I'll have to code in *8 into my extensions.conf instead of relying on the default built in 'steal' ? -- Chris - Original Message - From: Mimmus [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com; [EMAIL PROTECTED] Sent: Monday, March 20, 2006 1:17 PM Subject: RE: [Asterisk-Users] pickup problem PickUp2: http://linux.thorsten-knabe.de/asterisk/pickup.jsp works very well. Mimmus -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Panton Sent: Monday, March 20, 2006 4:50 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] pickup problem On 20 Mar 2006, at 15:39, Rich Adamson wrote: Mimmus wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk- [EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Monday, March 20, 2006 4:06 PM there is also a more generic call pickup using 'callgroup=2' and 'pickupgroup=2' in your sip definitions. That approach uses *8 or *8# to pickup any ringing phone within the callgroup number (eg, 2 in this example). Does this call pickup work with IAX2? If yes, how, if there is no callgroup/pickupgroup setting in iax.conf? More in general: does call pickup work between different protocols? Never had a need to do pickup with iax, so don't have a clue. As I recall, the callgroup keyword only applies to sip and zap channels. It doesn't work between protocols. Tim Panton ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Phones were working fine - Now there is noaudiowhen calling between extensions
On Tuesday 21 March 2006 10:55, C F wrote: Polycoms are not the best if you want a phone that works behind NAT. Are you kidding me? I used to think that anything SIP was a pain behind NAT until I turned on 'nat=yes' for the peer/user/friend entry in sip.conf and told the IP501 to register to the Asterisk box. (All defaults too, no special hyper-fast register interval or goofy Polycom configuration at all.) And even after that I wouldn't believe it until I had three of them behind a plain-jane WRT54G on standard Telco dynamic IP ADSL talking to my Asterisk box connected through third-party ADSL. Calls go out, calls come in, it's as if they're on the same LAN. Seriously: It Just Works. I keep popping into #asterisk-dev and thanking OEJ. I'm still not a huge fan of SIP but I have had *no* issues with Polycom IP501s behind NAT talking to an Asterisk box on a real IP. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] hfc-pci cards on ppc
is where anyone out there having hfc-pci cards running with asterisk on ppc-platform ? any information on working cards, drivers, kernel, asterisk versions would be helpful ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Native MOH - Convert mp3 to ulaw
I'd like to use native moh instead of with mpg123... for some reason the processes never bloody die. For native moh to not spawn an external player, I'd need to convert the default supplied moh sound files in /var/lib/asterisk/mohmp3 to ulaw and g729 format. Anyone know of a free, easy way to convert them? Thanks, Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] WiFi phones and WDS (Wireless Distribution System)
I'm about to start working with WiFi phones on my Asterisk installations. Can anyone tell me if they are using WiFi phones on wireless network that is extended with WDS and how well the phone handles jumping from access point to access point while on a call? Do any WiFi phones support WPA encryption or are they all still under the impression they are only being used on public hot spots? Thanks! Chip Schweiss ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Users Mailing List Traffic
Whether or not a forum is a better idea isn't really depending on the subject matter IMHO. Its success or failure depends on what the prospective participants like better. I personally cannot stand forums. That's a place where I have to expend energy to go there and manually click through stuff. If I remember to go there and say up to date, that is. Email comes to me, and is sorted suitably on the server side so there is no clutter. Deleting messages I don't care about is much easier than clicking myself through some thread on a forum. You never heard of a forum that sends new posts to you via email? I prefer forums where I can subscribe to the forum topics that interest me, and see only posts for those topics - yes, in my email. Then each message from the forum should have links to the CENTRALIZED FAQs (I understand there are a lot of different forums/faq's out there). That said, I seem to be in the minority in preferring forums for supprt related things like this - especially high volume stuff - so I'll just pipe down now... :) -- Best regards, Charles ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco POS 3-08-2
Anyone have experience with the 3-08-2 release of Cisco's SIP firmware? Are there any new features in the SIPDefault.cnf? Thanks, Ron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to make groups of extensions ???
Hi, I disagree that contexts are not for outgoing calls, how else do you restrict certain user to local calls only without using contexts?? On the subject of grouping extensions I use pickup groups so that any person can answer any phone in their immediate area by using a '*8' (as long as they belong to that group and they have the same context). Thanks Andrew Kohlsmith wrote: On Tuesday 21 March 2006 11:28, C F wrote: I disagree that it is very easy to miss, in fact even just mentioning it, makes it very easy to not miss, becuase it's a very understandable feature. Well when you know what to look for everything is easy to find. :-) How will groups without context help him? it's actualy both that he needs, however he will be able to get by without groups, but not without contexts. *nothing* works without contexts, which is why I said the answer doesn't help. Contexts are for incoming calls, not outgoing ones. How do contexts help him? -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to make groups of extensions ???
On Tuesday 21 March 2006 11:28, C F wrote:
I disagree that it is very easy to miss, in fact even just mentioning
it, makes it very easy to not miss, becuase it's a very understandable
feature.
Well when you know what to look for everything is easy to find. :-)
How will groups without context help him? it's actualy both that he
needs, however he will be able to get by without groups, but not
without contexts.
*nothing* works without contexts, which is why I said the answer doesn't help.
Contexts are for incoming calls, not outgoing ones. How do contexts help
him?
Okay, enough... ;)
For the OP, here's what needs to be done.
In sip.conf
[] ; user in Group A
whatever ip definitions are appropriate
context=FromGroupA
[] ; user in Group A
whatever ip definitions are appropriate
context=FromGroupA
[] ; user in Group B (the Boss)
whatever ip definitions are appropriate
context=FromGroupB
In extensions.conf
[FromGroupA]
whatever outgoing dialplan is appropriate
exten = _1888NXX,1,Dial(Zap/g1/${EXTEN})
[FromGroupB]
exten = _1888NXX,1,Dial(Zap/g4/${EXTEN})
[IncomingZap1]
whatever incoming dialplan is appropriate
[IncomingZap4]
whatever incoming dialplan is appropriate for the Boss
exten = s,1,Dial(SIP/,15,r) ; ring the Boss's phone
In zapta.conf
; first zap channel
whatever zap statements are appropriate
context=IncomingZap1
group=1
channel = 1
context=IncomingZap1
group=1
channel = 2
context=IncomingZap4
group=4
channel = 4
I did not try to actually implement the above statements, so syntax
might not be correct. However, the above example should be sufficient to
understand that incoming calls on line 4 are routed only to the Boss,
and outgoing calls by the Boss use the FromGroupB dialplan statements
(not the FromGroupA dialplan entries).
That should be more then enough to address the OP's original six
postings relative to both incoming and outgoing calls designated for
certain zap lines.
___
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Re: [Asterisk-Users] Phones were working fine - Now there is noaudiowhen calling between extensions
I didn't say it doens't work, I said it's not the best, and if you want I'll repeat myslef, Polycoms are not the best behind NAT, Cisco, or SPAs are much better. Just because you didn't run into any problems doesn't mean that it works well with all NAT devices. On 3/21/06, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Tuesday 21 March 2006 10:55, C F wrote: Polycoms are not the best if you want a phone that works behind NAT. Are you kidding me? I used to think that anything SIP was a pain behind NAT until I turned on 'nat=yes' for the peer/user/friend entry in sip.conf and told the IP501 to register to the Asterisk box. (All defaults too, no special hyper-fast register interval or goofy Polycom configuration at all.) And even after that I wouldn't believe it until I had three of them behind a plain-jane WRT54G on standard Telco dynamic IP ADSL talking to my Asterisk box connected through third-party ADSL. Calls go out, calls come in, it's as if they're on the same LAN. Seriously: It Just Works. I keep popping into #asterisk-dev and thanking OEJ. I'm still not a huge fan of SIP but I have had *no* issues with Polycom IP501s behind NAT talking to an Asterisk box on a real IP. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Phones were working fine - Now there is noaudiowhen calling between extensions
Andrew Kohlsmith wrote: On Tuesday 21 March 2006 10:55, C F wrote: Polycoms are not the best if you want a phone that works behind NAT. Are you kidding me? I used to think that anything SIP was a pain behind NAT until I turned on 'nat=yes' for the peer/user/friend entry in sip.conf and told the IP501 to register to the Asterisk box. (All defaults too, no special hyper-fast register interval or goofy Polycom configuration at all.) And even after that I wouldn't believe it until I had three of them behind a plain-jane WRT54G on standard Telco dynamic IP ADSL talking to my Asterisk box connected through third-party ADSL. Calls go out, calls come in, it's as if they're on the same LAN. Seriously: It Just Works. I keep popping into #asterisk-dev and thanking OEJ. I'm still not a huge fan of SIP but I have had *no* issues with Polycom IP501s behind NAT talking to an Asterisk box on a real IP. Same here with IP600; they just work. :) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to make groups of extensions ???
On 3/21/06, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Tuesday 21 March 2006 11:28, C F wrote: I disagree that it is very easy to miss, in fact even just mentioning it, makes it very easy to not miss, becuase it's a very understandable feature. Well when you know what to look for everything is easy to find. :-) How will groups without context help him? it's actualy both that he needs, however he will be able to get by without groups, but not without contexts. *nothing* works without contexts, which is why I said the answer doesn't help. Contexts are for incoming calls, not outgoing ones. How do contexts help him? Of course contexts are for outgoing as well, how else is he going to make sure that device a only dials out using channel/group x? -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WiFi phones and WDS (Wireless Distribution System)
There are a number of phones that support WPA, including the UTStarcom F1000G and F3000, and the Linksys WIP300 and WIP330, although current availability on these products is scarce. I know a couple of integrators that are having good success on WIFI deployments using the D-Link DWS-1008 (8 port Wireless Switch with PoE) and the corresponding DWL-8220AP Access points. In this scenario, administration of the AP's is handled through a central point on the switch. Here are a few specs on the switch. The D-Link MobileLAN solution is powered by Trapeze Networks and executes Trapeze Networks' Mobility System Software (MSS), which maintains the intelligence of the MobileLAN system. In addition to managing users' identities as they roam, the DWS-1008 configures and controls all aspects of the complementing DWL-8220AP Wireless Switch Dualband Access Points. Product Features: a.. Powered by Trapeze NetworksT Mobility System b.. Provides Central Management for WLAN Infrastructure c.. Automatically Configures All Attached DWL-8220APs AAA Authentication Offloading Capability The MobileLAN DWS-1008 supports Administration, Authorization, and Authentication (AAA) policies to ensure maximum security. Rather than checking the identity of a connecting user from the switch's local database, user authentication policies can be sent back to an AAA server for complete verification. This offloading capability ensures that the WLAN will not overload when clients are simultaneously connecting to the network. User-Based Authentication Services This wireless switch delivers Identity-based Networking, which provides user-based services such as virtual private group membership, personal firewall filters, time-of-day/day-of-week access, encryption type, authentication, usage tracking, location tracking, and associated network statistics. Authorizations stay with users wherever they roam because all deployed DWS-1008s share stored information, ensuring secure access and connectivity to the right services. Easy Deployment The DWS-1008 includes eight 10/100 Mbps ports with integrated PoE to enable network connectivity to any connected DWL-8220AP. It is designed for distributed deployments in the wiring closet or small or medium offices. It can support up to six directly connected DWL-8220APs and up to six more DWL-8220APs connected indirectly. Maximum Performance With Load Balancing Capabilities The DWS-1008 performs Layer 2 forwarding and also comes with extensive Layer 3-4 and identity-tracking capabilities. It integrates seamlessly with wired infrastructures and offers redundant load-sharing links, 802.1q trunking, spanning tree and per-VLAN spanning tree (PVST+). It also supports IGMP snooping, which is vital to supporting IP multicast streams. Cory Andrews ++ VOIPSupply.com A Division of b2 Technologies 454 Sonwil Drive Buffalo, NY 14225 direct - 716.250.3402 mobile - 716.907.4059 email - [EMAIL PROTECTED] AIM - b2Cory - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, March 21, 2006 12:02 PM Subject: [Asterisk-Users] WiFi phones and WDS (Wireless Distribution System) I'm about to start working with WiFi phones on my Asterisk installations. Can anyone tell me if they are using WiFi phones on wireless network that is extended with WDS and how well the phone handles jumping from access point to access point while on a call? Do any WiFi phones support WPA encryption or are they all still under the impression they are only being used on public hot spots? Thanks! Chip Schweiss ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WiFi phones and WDS (Wireless Distribution System)
Hi, I use the Zyxel P-2000W v2 wireless VOIP phones with Zyxel G-1000 access points and the hand off calls fairly smoothly using a port for the hand off and using WEP security (the Zyxel is not capable of WPA security yet). I understand that people have problems with some manufactures access points not handling the hand off very well due to latency issues. I can not remember the article but I believe Network World or a similar rag did hand off tests and found the Zyxel to be one of the best at the time. Thanks [EMAIL PROTECTED] wrote: I'm about to start working with WiFi phones on my Asterisk installations. Can anyone tell me if they are using WiFi phones on wireless network that is extended with WDS and how well the phone handles jumping from access point to access point while on a call? Do any WiFi phones support WPA encryption or are they all still under the impression they are only being used on public hot spots? Thanks! Chip Schweiss ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Realtime 1.2.5 and Username/auth name mismatch ?
Hello, I installed 1.2.5 and realtime SIP. The connection to the DB is OK because I can get the values from the CLI. Here are my 3 different cases: 1- If I put an unexisting user, I get 404 and I am not able to dial. 2- If I check "Disable registration" within Firefly itdoes notregisterbut I am able to dial a destination (...) 3- If I leave registration ON, I get the 404 messagebut I amnot ableto dial a destination This is weird, anyone had this before ?? :- ) Thanks in advance ! Frederic Mar 21 14:11:36 NOTICE[894]: chan_sip.c:10854 handle_request_register: Registration from '""sip:[EMAIL PROTECTED]:5060;transport=udp' failed for '192.168.1.5' - Username/auth name mismatch SNET-PBX*CLI realtime mysql statusConnected to [EMAIL PROTECTED], port 3306 with username asterisk for 1 minutes, 35 seconds. SNET-PBX*CLI realtime load sipusers username Column Name Column Value id 1 name accountcode callerid canreinvite no context internal defaultip 0.0.0.0 host dynamic insecure very language br nat yes port 0 qualify no secret type friend username disallow all allow g729 allow ilbc allow gsm allow ulaw allow alaw regseconds 0 ipaddr 0.0.0.0 cancallforward yes ++--+--+-+-++| Field | Type | Null | Key | Default | Extra |++--+--+-+-++| id | int(11) | | PRI | NULL | auto_increment || name | varchar(80) | | UNI | | || accountcode | varchar(20) | YES | | NULL | || amaflags | varchar(13) | YES | | NULL | || callgroup | varchar(10) | YES | | NULL | || callerid | varchar(80) | YES | | NULL | || canreinvite | char(3) | YES | | yes | || context | varchar(80) | YES | | NULL | || defaultip | varchar(15) | YES | | NULL | || dtmfmode | varchar(7) | YES | | NULL | || fromuser | varchar(80) | YES | | NULL | || fromdomain | varchar(80) | YES | | NULL | || fullcontact | varchar(80) | YES | | NULL | || host | varchar(31) | | | | || insecure | varchar(4) | YES | | NULL | || language | char(2) | YES | | NULL | || mailbox | varchar(50) | YES | | NULL | || md5secret | varchar(80) | YES | | NULL | || nat | varchar(5) | | | no | || deny | varchar(95) | YES | | NULL | || permit | varchar(95) | YES | | NULL | || mask | varchar(95) | YES | | NULL | || pickupgroup | varchar(10) | YES | | NULL | || port | varchar(5) | | | | || qualify | char(3) | YES | | NULL | || restrictcid | char(1) | YES | | NULL | || rtptimeout | char(3) | YES | | NULL | || rtpholdtimeout | char(3) | YES | | NULL | || secret | varchar(80) | YES | | NULL | || type | varchar(6) | | | friend | || username | varchar(80) | | | | || disallow | varchar(100) | YES | | all | || allow | varchar(100) | YES | | g729;ilbc;gsm;ulaw;alaw | || musiconhold | varchar(100) | YES | | NULL | || regseconds | int(11) | | | 0 | || ipaddr | varchar(15) | | | | || regexten | varchar(80) | | | | || cancallforward | char(3) | YES | | yes | || setvar | varchar(100) | | | | |++--+--+-+-++ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Users Mailing List Traffic
Charles Marcus wrote: Whether or not a forum is a better idea isn't really depending on the subject matter IMHO. Its success or failure depends on what the prospective participants like better. I personally cannot stand forums. That's a place where I have to expend energy to go there and manually click through stuff. If I remember to go there and say up to date, that is. Email comes to me, and is sorted suitably on the server side so there is no clutter. Deleting messages I don't care about is much easier than clicking myself through some thread on a forum. You never heard of a forum that sends new posts to you via email? I prefer forums where I can subscribe to the forum topics that interest me, and see only posts for those topics - yes, in my email. Then each message from the forum should have links to the CENTRALIZED FAQs (I understand there are a lot of different forums/faq's out there). That said, I seem to be in the minority in preferring forums for supprt related things like this - especially high volume stuff - so I'll just pipe down now... :) can't forget about forum RSS feeds too.. then you can selectively pick which messages you want to view.. -- . -BEGIN GEEK CODE BLOCK- Version: 3.1 GCS/GIT d-@ s+:+ a? C+++ BLHIS$ U+++ P+ L+++ !E W+++$ N++ o+ K w-- PS+++ PE@ Y+ PGP++ t 5? X !R tv+ b- DI-- D G e+ h r+++ y --END GEEK CODE BLOCK-- . signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to make groups of extensions ???
On Tuesday 21 March 2006 12:07, Chuck Bunn wrote: I disagree that contexts are not for outgoing calls, how else do you restrict certain user to local calls only without using contexts?? On Quite simply: The SIP user has a 'context' field. When a call comes *in* to Asterisk from that SIP user, they get dumped into the specified context. Contexts are for incoming calls, not outgoing. You don't get to specify a context on for a peer, only a user/friend. the subject of grouping extensions I use pickup groups so that any person can answer any phone in their immediate area by using a '*8' (as long as they belong to that group and they have the same context). Yes absolutely. That's for incoming though, the original poster was trying to restrict which trunk lines each extension could access for outgoing calls. This is done with 'group=# and Dial(Zap/[gGrR]#) and a little dialplan magic, or with something I'm not able to see which the other poster is hinting at. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to make groups of extensions ???
Yeah, I agree with Chuck. User's on our system are put into various contexts depending on who they can call... local, long distance, or internal only. Aaron On Tue, 21 Mar 2006, Chuck Bunn wrote: Hi, I disagree that contexts are not for outgoing calls, how else do you restrict certain user to local calls only without using contexts?? On the subject of grouping extensions I use pickup groups so that any person can answer any phone in their immediate area by using a '*8' (as long as they belong to that group and they have the same context). Thanks Andrew Kohlsmith wrote: On Tuesday 21 March 2006 11:28, C F wrote: I disagree that it is very easy to miss, in fact even just mentioning it, makes it very easy to not miss, becuase it's a very understandable feature. Well when you know what to look for everything is easy to find. :-) How will groups without context help him? it's actualy both that he needs, however he will be able to get by without groups, but not without contexts. *nothing* works without contexts, which is why I said the answer doesn't help. Contexts are for incoming calls, not outgoing ones. How do contexts help him? -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ISDN Protocol Unknom Error with Junghanns OctoBRI
I think that you are sending an outgoing caller id that is not part of the DID range. Most operators do not allow this. ISDN/Telephony Numbering Plan (E.164/E.163) (1) '1013' ] Are you using caller id 1013 ? Change it to a number that is part of your trunks. Henk -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sébastien Mortier Sent: maandag 20 maart 2006 11:52 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] ISDN Protocol Unknom Error with Junghanns OctoBRI Hello, I recently bought a Junghanns Octobri Card. I have some problems with this card to make outbound calls but I can receive calls. I have 3 lines to PSTN and 3 lines to my existing PBX FRANCE TELECOM -- OctoBRI -- Asterisk 1.2.4-BRIstuffed-0.3.0-PRE-1h -- OctoBRI -- PABX e-Generis ISDN Phones | | SIP Phones France Telecom -- SIP Phones : Works France Telecom -- ISDN Phones : Works SIP Phones -- ISDN Phones : Works ISDN Phones - SIP Phones : Works SIP Phones -- France Telecom : DOESN'T WORK ISDN Phones - France Telecom : DOESN'T WORK Here are some characteristics of my Asterisk Setup OS Linux Gentoo 2.6.15-r1 zaptel 1.2.3 libpri 1.2.2 asterisk 1.2.4-BRIstuffed-0.3.0-PRE-1h ISDN Lines : EuroISDN not EuroISDN+ Junghanns OctoBRI PCI ISDN Card S/T 1+8 - S/T 2+7 : TE Mode S/T 3+6 - S/T 4+5 : NT Mode modprobe qozap ports=60 zaptel.conf --- loadzone=fr defaultzone=fr # qozap span definitions # most of the values should be bogus because we are not really zaptel span=1,1,3,ccs,ami span=2,0,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami span=5,1,3,ccs,ami span=6,0,3,ccs,ami span=7,0,3,ccs,ami span=8,0,3,ccs,ami bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12 bchan=13,14 dchan=15 bchan=16,17 dchan=18 bchan=19,20 dchan=21 bchan=22,23 dchan=24 --- zapata.conf --- switchtype = euroisdn pridialplan = dynamic prilocaldialplan = local nationalprefix = 0 internationalprefix = 00 usecallingpres = yes echocancel = yes echocancelwhenbridged = yes echotraining = 100 callprogress=yes context=isdn-incoming group = 1 ; S/T port 1,2,7,8 channel = 1-2 channel = 4-5 ;channel = 19-20 channel = 22-23 context=pbx-incoming group = 2 channel = 7-8 channel = 10-11 ;channel = 13-14 channel = 16-17 - Here's the output BRI debug when I try to make outbound calls from a SIP phone : -- Executing Dial(SIP/400-c8dc, Zap/1/1013) 1 -- Making new call for cr 137 -- Requested transfer capability: 0x00 - SPEECH 1 Protocol Discriminator: Q.931 (Cool len=26 1 Call Ref: len= 1 (reference 9/0x9) (Originator) 1 Message type: SETUP (5) 1 [1 041 031 801 901 a31 ] 1 Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) 1 Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) 1 Ext: 1 User information layer 1: A-Law (35) 1 [1 181 011 811 ] 1 Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Preferred Dchan: 0 1 ChanSel: B1 channel 1 ] 1 [1 6c1 051 411 801 341 301 301 ] 1 Calling Number (len= 7) [ Ext: 0 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) 1 Presentation: Presentation permitted, user number not screened (0) '400' ] 1 [1 701 051 c11 311 301 311 331 ] 1 Called Number (len= 7) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '1013' ] -- Called 1/1013 1 Protocol Discriminator: Q.931 (Cool len=8 1 Call Ref: len= 1 (reference 137/0x89) (Terminator) 1 Message type: RELEASE COMPLETE (90) 1 [1 081 021 871 e41 ] 1 Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: International network (7) 1 Ext: 1 Cause: Unknown (100), class = Protocol Error (6) ] 1 -- Processing IE 8 (cs0, Cause) -- Channel 0/1, span 1 got hangup, cause 100 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null 1 NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null -- Hungup 'Zap/1-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing Hangup(SIP/400-c8dc, ) == Spawn extension (default, 1013, 2) exited non-zero on 'SIP/400-c8dc' 1 received TEI check request for TEI = 127 I've already tested several configurations for zapata.conf especially with the pridialplan and switchtype lines but without success. Could you help me to analyse and solve this odd problem ? Thank you in advance, -- Sébastien Mortier AbsysTech Tel : +33 3 20 50 99 02 Fax : +33 3 20 74 50 05 Gsm : +33 6 20 79 24 29 http://www.absystech.fr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by
[Asterisk-Users] VoIP prepaid billing
Hi, I am using the following architecture: SER (SIP server) -- Asterisk -- PSTN Gateway And I would like to implement a prepaid billing solution that could be controlled by asterisk. Can anyone give me a direction? Jose Simoes ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Aterisk with Realtime
the DB in same server but iam using DSN to connect using ODBC is the not th right proces if not kindly recomend me the process i want to both SIP users / CDR to be from Mysql ram On 3/21/06, Aaron Daniel [EMAIL PROTECTED] wrote: Are you moving your db over to an odbc connection?AaronOn Tue, 21 Mar 2006, ram wrote: Hi i have installed unixODBC Drivers and created DSN but when i reload from * console asterisk terminating the services if i do again start the services and change something and reload again its hangs is that bug in 1.2.5 ram On 3/21/06, Aaron Daniel [EMAIL PROTECTED] wrote: Since you say you're using mysql as the backend, you need to change anything that says odbc to mysql so that the server knows where to find the db at.Also, you need to make sure the DB info is in res_mysql.conf. Aaron ram wrote: Hi thanks for the reply this what my extconfig sipusers = odbc,asterisk,2_Sip sippeers = odbc,asterisk,2_Sip extensions = odbc,asterisk,2_Extensions voicemail= odbc,asterisk,2_VMUsers voicemail_messages = odbc,asterisk,2_VM waht is wrong with this ? ram On 3/20/06, *Aaron Daniel* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: That, and make sure you've got extconfig set to use mysql for it's sippusers and sippeers and not odbc. Aaron Patrick wrote: On Mon, 2006-03-20 at 23:14 +0530, ram wrote: Hi iam working with asterisk with mysql Realtime when i have confgured and run the asterisk iam getting the following error i dig all the places for help could not find the results could some one help me what is wrong iam using 1.2.5 on FC4 Mar 20 23:04:52 NOTICE[2054] cdr.c: CDR simple logging enabled. Mar 20 23:04:52 NOTICE[2054] indications.c: Removed default indication country 'us' Mar 20 23:04:52 WARNING[2054] pbx_config.c: The use of '_.' for an extension is strongly discouraged and can have unexpected b ehavior.Please use '_X.' instead at line 3 Read the message and do as it suggests: in your dialplan replace all _. with _X. Patrick ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users--Aaron DanielComputer Systems TechnicianSam Houston State University[EMAIL PROTECTED] (936) 294-4198___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to make extension groups ???
At 01:50 AM 03/21/2006, you wrote:
i am repeating this question for the sixth time but i think i was
not explaining the problem correctly. . Now i will try
to explain it..
For each line in Zapata.conf make an entry like:
context=line1
group=1,9
channel = 1
context=line2
group=2,9
channel = 2
context=line3
group=2,3,9
channel = 3
context=line4
group=2,3,4,9
channel = 4
Then in sip.conf assign the extensions to the outgoing contexts as
shown in extensions.conf:
[12]
context=group2
[13]
context=group2
[14]
context=group3
[20]
context=group3
[21]
context=group4
[30]
context=group4
Then in extensions.conf make your dial entries look something like:
[group1]
include = emergency
exten = _x.,1, dial(ZAP/g1, ${EXTEN})
[group2]
include = emergency
exten = _x.,1, dial(ZAP/g2, ${EXTEN})
[group3]
include = emergency
exten = _x.,1, dial(ZAP/g3, ${EXTEN})
[group4]
include = emergency
exten = _x.,1, dial(ZAP/g4, ${EXTEN})
[emergency] 911,1,dial(ZAP/g9, 911)
That's the tip of the iceberg. It's everything you need to know if
you've read the documentation about setting up *.
Ira
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Re: [Asterisk-Users] WiFi phones and WDS (Wireless Distribution System)
On 3/21/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I'm about to start working with WiFi phones on my Asterisk installations. Can anyone tell me if they are using WiFi phones on wireless network that is extended with WDS and how well the phone handles jumping from access point to access point while on a call? Do any WiFi phones support WPA encryption or are they all still under the impression they are only being used on public hot spots? We've got a WDS mesh setup on WRT54GS's here internally with the HyperWRT firmware, and we're able to roam fairly seamlessly with the Linksys WIP300 and the Zyxel phones. The F1000 from UTStarcom gets a little cranky while you're on a call and go from one to the other. The Linksys WIP300 does support WPA and 802.11g. We got all of the aforementioned equipment from VoipSupply.com. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ODBC and VoiceMail messages.
Is it possible to store voicemail recorded messages using odbc? Fernando Lujan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF leak with IAXy call waiting Bug?
I have only been able to confirm this issue with my IAXy as it is the only ATA I have. I am running 1.2.5 stable. Example: User @ exten 100 (iaxy) receives call from PSTN (call 1). While on the call, another user from PSTN (call 2) calls 100 (iaxy) sending a call waiting beep to the iaxy. User 100 answers the call from PSTN call 2 putting PSTN call 1 on hold with music. While call 1 is on hold, the user hits DTMF buttons on their phone and user @ 100 hears these DTMF tones. This should not happen as this user (call 1) is on hold and shouldn't be able to interupt the call between 100 and call 2. Can someone else try and re-produce this and see if this is a bug? Thank you! Zachary McGibbon [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with chan_iax.c implimentation causes bad audio?
On 21 Mar 2006, at 13:21, Rich Adamson wrote: Add to that the fact that iax is actually a proprietary protocol implementation (eg, not based on any current published/approved standards), and the fact that only folks that run asterisk actually use the protocol, you now have a fairly major support issue from the itsp's perspective. There is an RFC draft, the latest version is complete enough to implement IAX2 without asterisk source. I know it isn't a standard, but it is published. Tim. Tim Panton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with chan_iax.c implimentationcausesbadaudio?
On 21 Mar 2006, at 16:19, Adam Robins wrote: All switches and routers give highest priority to traffic on IAX2 port 4569. We use DSCB values over the IP-VPN to prioritize it as well. This did not change with the upgrade, as we can still see proper packet coding. The softphone is provided by our vendor Aheeva. It is the same IAX2 softphone they use in their own call centers. Funny thing is that they say that moving to Asterisk 1.2.4 tremendously IMPROVED their call quality with IAX2. Headsets are Plantronics H251N tops with DA60 USB adapters. All Desktops are at least 2.0 GHz P4 with 512MB RAM I don't suppose you have an ethereal packet capture from a bad call ??? Or a description of the 'badness'? I'm doing stuff in IAX2 at the moment and might be able to spot a problem. Tim. Tim Panton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Realtime / SIP Peers etc
Ready to scream here.. 1. After 6 months with Asterisk I'm STILL trying to understand the difference between a SIP user, friend and peer. 2. Exactly what resource does Asterisk use to send MWI to registered phones? I thought it was astdb? 3. It looks like it isn't astdb. It looks like it will only send MWI to a phone if it shows up in 'sip show peers'. 4. WHY then does a reload clear this list? Doesn't this list come from the astdb file? 5. Why is this such a damn mess? Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to make groups of extensions ???
On Tuesday 21 March 2006 12:25, Aaron Daniel wrote: Yeah, I agree with Chuck. User's on our system are put into various contexts depending on who they can call... local, long distance, or internal only. And *all* of those people are placing calls *in* to asterisk to get into those contexts. :-) When you pick up a telephone wired into an FXS port; asterisk sees an incoming request for dialtone. When you pick up your SIP phone and dial; it must match a friend or user entry or you'll never get in. When your IAX softphone client makes a call, again, it must match a friend or user entry. These are *all* incoming calls as far as Asterisk is concerned. You get dumped into a specific part of the dialplan (the context specified) and you tell Asterisk what they can dial. Internal extensions, external peers, Zap channels or even applications... the second half of all of this is the outgoing part, when Asterisk Dial()s. -A. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to make groups of extensions ???
On Tuesday 21 March 2006 12:14, C F wrote:
Of course contexts are for outgoing as well, how else is he going to
make sure that device a only dials out using channel/group x?
No, the dialplan determines what you do. I.e. you get to an appropriate
Dial() command which specifies the appropriate group.
There's absolutely nothing stopping someone from writing this:
[context-1]
exten = 123,Dial(Zap/g1/${EXTEN})
exten = _X.,1,Goto(context-2,${EXTEN},1)
[context-2]
exten = _X.,1,Dial(Zap/g2/${EXTEN})
Your contexts didn't determine a thing; the Zap group determined it for your
outgoing call.
(I *do* see what you're saying, but honestly the context has absolutely
nothing to do with it short of dumping the extension into the correct part of
the dialplan. It's the group configuration that does the only dial out
through line 4,5,6.)
-A.
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Re: [Asterisk-Users] ISDN Protocol Unknom Error with Junghanns OctoBRI
On Mon, 2006-03-20 at 11:51 +0100, Sébastien Mortier wrote: Hello, I recently bought a Junghanns Octobri Card. I have some problems with this card to make outbound calls but I can receive calls. I have 3 lines to PSTN and 3 lines to my existing PBX FRANCE TELECOM -- OctoBRI -- Asterisk 1.2.4-BRIstuffed-0.3.0-PRE-1h -- OctoBRI -- PABX e-Generis ISDN Phones | | SIP Phones France Telecom -- SIP Phones : Works France Telecom -- ISDN Phones : Works SIP Phones -- ISDN Phones : Works ISDN Phones - SIP Phones : Works SIP Phones -- France Telecom : DOESN'T WORK ISDN Phones - France Telecom : DOESN'T WORK zaptel.conf --- loadzone=fr defaultzone=fr # qozap span definitions # most of the values should be bogus because we are not really zaptel span=1,1,3,ccs,ami span=2,0,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami span=5,1,3,ccs,ami span=6,0,3,ccs,ami span=7,0,3,ccs,ami span=8,0,3,ccs,ami I was going mad with a Quadbri until I changed ccs,ami to ccs,hdb3 and it's been running 6 months now. -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ODBC and VoiceMail messages.
On Tue, 21 Mar 2006 12:56:13 -0500, Fernando Lujan [EMAIL PROTECTED] wrote: Is it possible to store voicemail recorded messages using odbc? Fernando Lujan see asterisk-sources/doc/README-odbcstorage -- Justin Tunney ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problem with chan_iax.cimplimentationcausesbadaudio?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Tuesday, March 21, 2006 11:36 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Problem with chan_iax.cimplimentationcausesbadaudio? On Tuesday 21 March 2006 11:19, Adam Robins wrote: All switches and routers give highest priority to traffic on IAX2 port 4569. We use DSCB values over the IP-VPN to prioritize it as well. This did not change with the upgrade, as we can still see proper packet coding. Right, I wouldn't suspect otherwise. The softphone is provided by our vendor Aheeva. It is the same IAX2 softphone they use in their own call centers. Funny thing is that they say that moving to Asterisk 1.2.4 tremendously IMPROVED their call quality with IAX2. I wonder what the hell is going on then, that is definitely something strange. Headsets are Plantronics H251N tops with DA60 USB adapters. All Desktops are at least 2.0 GHz P4 with 512MB RAM Thanks for the information. I feel bad for not having a good solid answer for why it's occurring. As the saying goes: I don't have an answer, but I admire the problem... -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] need to make my oh323 work with quintum no gatekeeper
Hi all, Can someone share with me his experience in making asterisk-oh323 talk to quintum gateway without gatekeeper. My set up is QUINTUM GATEWAY --IPM ASTERISK (OH323) Both are gateways.. but I don't know what authentication I will set up in oh323.conf and how to set it up I will be glad if anyone can help Goksie ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] need to make my oh323 work with quintum no gatekeeper
Hi all, Can someone share with me his experience in making asterisk-oh323 talk to quintum gateway without gatekeeper. My set up is QUINTUM GATEWAY --IPM ASTERISK (OH323) Both are gateways.. but I don't know what authentication I will set up in oh323.conf and how to set it up I will be glad if anyone can help Goksie ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 计划生育的无耻宣传 该结束了
这个名单是英文. 这是我讲的一切. Jeffery Chen wrote: 真的很遗憾。不管左派的网友还是右派的网友，在谈到计划生育的时候大都会摆出 一副冷酷的面孔。我就来说说计划生育是个什么东西。 坐在电脑面前的精英们应该知道这么一个国情常识：中国的农民是没有任何退 休金和任何形式的医疗保障的。 你们有没有想过，他们如果没有一个强有力的孩子，当他们失去劳动能力的时 候，就只能坐在家里慢慢饿死？死并不可怕，对于中国农民来说，每年的非正常死 亡不计其数：有死在矿井里的，有死在城市的工地上的，有死在收容所里的，有洪 水淹死的，有吞农药自杀的，有上访的时候跳楼的，当然也有死在强制堕胎的病床 上的。这些都不能让农民恐惧，为什么？因为他们总怀着一线希望能逃过这些磨 难，他们一生都以极高的热情在和这些死的可能性做斗争。但是有一种磨难是不可 能逃过去的，那就是衰老。 如果一个农民在只有一个女儿的情况下被结扎了，那么就意味着他在很年轻的 时候就已经预见到了自己的晚年：除非自杀，否则就只能在极度的物质匮乏中衰竭 而死，失去劳动能力的一天就是他们的死期。这种对死亡的确切的预期是多么恐怖 你们想过么？一个人在年轻的时候就能预见到自己怎么死，这好玩不好玩呢？ 当然，中国农民的职业寿命也是非常之长的。在网上有很多图片都是70岁以上 的老人在背柴火或者乞讨，他们算是很幸运的，自己尚能够老有所用养活自己， 但是他们很清楚等待他们的将是什么：生命中注定没有一天的假期，退休的日子便 是他们的生活来源彻底枯竭的日子。 有人开始从理论上做分析了。即使生的是女儿也有赡养老人的义务啊！就算嫁 出去了，她和他老公的财产是双方共有的呀！您要是这么想，就麻烦你到农村看一 看吧。农民并不是都不尊重自己的老婆，我也见过感情好的。但是女性对家庭财产 的支配权真的是微乎其微，想把种地挣的一点钱拿回娘家去给女方的父母花？看着 吧，老公的棍子就要下来了。 我也想过，如果女婿不承担赡养义务是不是可以打官司呢？可是稍微动一下脑 子就知道，这根本是不可能的。连给爹妈的钱都没有，难道能有交诉讼费的钱么？ 我认识一个打过离婚官司的小时工，她曾告诉我，他老公威胁法官：只要你判我离 婚，我就砍你全家。当然，我举这个例子决不是想说明基层的法官好欺负，他们可 不是善主。只是面对一无所有只有烂命一条的百姓，他们是不愿意拼命的。 中国的农村，想通过法律手段解决问题？想起来脑袋都大了。 应该说，极不健康极不公正的社会环境在逼迫农民生育，而凶残和腐败的地方 执法机构又在用各种方式制止农民生育。要减少人口，是应该助长后者还是结束前 者呢？或者说，是应该尽量逼迫不生育，还是应该尽量不逼迫生育呢？ 如果真的是为了人民的福利，这个国家有太多比计划生育更可行的方法了。但 是这些方法大人们时不屑于用的。 除了增加养老和医疗保障之外，还有一个最简单的办法，就是土地私有。在土 地国有的大环境下，农民除了孩子是一无所有的，只有生一个胖小子能让农民找到 一点拥有的感觉。 土地国家或集体所有的条件下，有一个永远无法解除的困境，就是土地如何分 配的问题。即使我们把农村的官老爷想象得无比清廉，无比公正，那么请问，当人 口发生变化的时候，他们如何公正地调整土地使用权呢？ 当然，各地都有不同的办法。我查了各种土地方面的法律，大多语焉不详（原 谅我没有做过什么乡土调查，没人给我报路费啊），但是总的来讲，还是以人口为 基准的。换句话说，农民多生孩子虽然要被罚款，但是在分配土地使用权的时候， 还是会有一些隐性的好处。 而土地私有以后的农民就不一样了，因为有了自己的财产，自己种不动了可以 出租，自然心里就塌实了。生多了孩子不仅不能带来什么好处，反而会因为劳动力 过剩而降低自家的生存质量。那么不用你计划，人家也自然会去限制生育。 土地私有对于大人们来说当然是不能接受的了。正是靠着对土地的所有权，国家把 人民牢固地掌握起来：因为你脚下的土地都是国家的，只要你不会飞，你就时时刻 刻地欠着国家的人情，因为你踩了它的地。正因为如此，无论城市还是农村的暴力 拆迁都显得那么理直气壮。 计划生育嘛，呵呵，正是这样一种和土地国有相辅相成的政策：国有的土地相 当于农场主的一个巨大的畜栏，被限制生育的人民像是被阉割的只能干活的牲畜。 这两种措施有效的让人民对国家的依附关系建立得天衣无缝。 以上只是说农民为什么要生的问题。还有就是，人口多究竟有没有那么可怕。 有人计算过新增的国民要吃掉多少GDP云云。我听了简直要喷。中国的农民确实是 劳动生产率低，这我承认，但是人家什么时候吃过别人创造的财富了？中国农民每 年要给国家上缴各种税费，而从来没有得到过一分钱的福利，每修一段破烂公路还 都要强行的集一次资！请问，他们消耗掉国家什么了？你们这些白领创造的GDP有 哪一分钱是进入了农民的腰包了。不会把你给你家保姆发的工资也算上吧？啊？没 人逼你雇保姆啊！ 恰恰相反，超生不但没有给国家带来负担，反而让地方政府有了更好的剥皮抽 筋的理由，计划生育官员就像大城市里的交通警和小城市里的扫黄警察一样，每天 都在期待着有人犯法，好来送钱给他们。 你们可以去设想未来中国的福利如何如何。但是在这个年代，社会福利对于户 口本上写着农业二字的人来说还是一个虚拟物品的时候，请不要去咒骂别人占用 你的GDP好不好？网上有的是中国底层的照片，你看人家哪个像是吃你们丫的GDP过 活的？有的冷酷并不是道德原因造成的，而是因为逻辑思维的缺乏，那就好好锻炼 一下你的逻辑思维。 有人提到超生导致的残障人口。避免先天性残障当然是任何一个政府都会做 的。但是，我还想提醒一下，中国大部分残障人士也是没有任何福利的呀！也是只 能家人养着的呀！即使是享受微弱福利的城市户口的残障人士，他们的数量也远没 有中国贪官污吏的数量多吧。而一个乡镇级贪官的开销（包括汽车、手机、吃喝、 嫖、旅游、盖办公楼、名牌烟酒、送子女去省城上学……）按一个月5000块算不多 吧？那就顶得上20个城市贫民的最低生活保障（也就是国家花在他们身上的所有的 钱）。至于县级？市级？省级？X级……的干部，一人顶1000个残疾人不在话下吧？ 计划生育和反贪也许并不截然矛盾。但是把计划生育上升到基本国策，分明就 是把国家落后的责任推卸给普通老百姓。如果有这么一个人，他在声色场所挥霍无 度，却在去菜市场买菜的时候讨价还价，你会不会觉得他有病？国家花那么多力量 来搞计划生育，正是这样一个有病的表现。 当然，中国经常干这种事情。比如希望工程吧，这么多年据说也就募到了20个 亿。你说好笑不好笑，国家随便少干一件蠢事不能省出20个亿？要让我们捐钱？为 什么要丢西瓜拣芝麻，这可能只有政策制定者自己心里清楚。要不大家都来猜一猜？ 然后，请允许我再往下说一层。 人到底是什么？是一个国家富强的手段，还是一个国家富强的目的？人口问 题？人口不是问题，人口不就是你和我构成的？人口不是国家豢养的牲口，需要耕 地或挤奶就多产一点，养活不了就少产一点。恰恰相反，人口是这个国家的主人， 国家要无条件服从人口的需要而不是相反。 如果一个国家的妇女要承受强迫结扎、强迫堕胎的痛苦，要被别人用暴力剥夺 自己腹中的胎儿，这个国家再富强又有什么意义？当妇女们被成群关在拘留所里， 警察等着她们一个个地签字同意结扎，然后直接用卡车拖到医院，这个国家作为一 个人类生活的地方还值得存在下去么？ 中国妇女当然从来没有过过好日子：一夫多妻、裹小脚、用生命保贞洁……但也 从没有像现在这样被剥夺了亚当夏娃时就有的伟大的生育的权力呀。 正是因为用考虑畜牧业的方式来考虑人口，把农民当成国家的财产而不是主 人，才会出现这样一个荒谬的情况：一方面总说人口多，一方面却无耻地限制老百 姓出境，对于基层老百姓办护照百般刁难！ 不是说人多么？为什么不让人家到别的国家去？为什么办护照还要审批？为什 么北京上海这些所谓的高素质人口出国反而不受限制，为什么农民跑出去就不 行？如果不是把人家当成田地里的劳动机器，还有什么其他原因呢？ 我想请大家看一条很少被注意的法律。这是《中华人民共和国出入境管理法实 施细则》中的一句话：出境就业，须提交聘请、雇用单位或者雇主的聘用，雇用证 明。这里的提交不是向负责签证的老外提交，而是向户口所在地的市、县公安 局出入境管理部门提交。如果按照某些大人们抱怨的那样，中国穷是因为人太 多，那应该积极鼓励大家出国打工才好。当然不要求领导们花时间去帮他们在国外 找工作，但至少不该限制人家。即使没有雇佣证明，人家出去以后再想办法又有 什么关系呢？ 别告诉我什么给国家丢脸。让贫苦的农民担负起给国家挣面子的责任是毫无 道理的。请问他们在这几十年的生活里，什么时候有过尊严可言？不能让占人口大 多数的农民过得高兴，这个国家还能有面子么？ 如果有人提出通过饿死一批人来减少人口，大家肯定不会同意。因为你们都知 道生存权是全世界公认的人权，甚至中国还把它说成是中国对人权理论的一大贡 献。但是通过限制生育甚至强迫结扎来减少人口，大家居然就认可了，也就是 说，一般人认为生育权没有生存权那么重要。 可是，你们知道么？对于任何一种正常的生物来说，生育都是比生存更加神圣 的使命。人也是不能例外的。对于没有宗教的民族尤其如此，因为只有基因的延续 能给人带来永恒体验。 我们来举一个例子。设想一个母亲有不止一个孩子，当其中一个孩子的生命受 到威胁的时候，你说她会不会用自己的生命去换取这个孩子的生命呢？我可以告诉 你们，99%的母亲都会这样的。为什么呢？这是所有能在进化大潮中保留下来的基 因共有的自我保护机制在起作用，它们在下意识中暗示着每个人：牺牲个体，让基 因延续下去。 不排除有能生育而不愿生育的人，就好像有能活下去但选择自杀的人一样。这 是另一回事。我现在说的是，对于想生育的人不允许其生育是多么的残忍。 凡是为计划生育基本国策叫好的人，请你们务必发发慈悲，看看农民的生活现 状。以这样奴隶般的生活质量，即使是纯粹为了高兴而生孩子也是毫不过分的。 这个国家欠农民的太多了，看在1960年前后那3000万冤魂的份上，别再折磨他 们了吧。 -- Jeffery iaxtel Num: 1-700-576-1311 fwdnet Num: 728150 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:
Re: [Asterisk-Users] WiFi phones and WDS (Wireless Distribution System)
the cisco 7920 with the latest firmware supports WPA-psk using the AKM for auth. it is important to turn off CDP discovery otherwise it will crash other cisco sccp phones connected to asterisk - advaned menu: Menu, *, #, #, Send (green phone icon) - network config and disable cdp tx haven't had a chance to roam with it just yet, but it works fine. j Cory Andrews wrote: There are a number of phones that support WPA, including the UTStarcom F1000G and F3000, and the Linksys WIP300 and WIP330, although current availability on these products is scarce. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problem with chan_iax.cimplimentationcausesbadaudio?
Thanks for the offer. We deleted all of our Ethereal traces once we switched to SIP. On a bad call call there were tens of thousands of checksum errors and packets out of sequence. This occurred both with and without IAX2 trunking and trunktimestamps. Complaints of poor quality were from both the agent and customer sides. Mostly cutting in and out - typical of dropped packets. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Panton Sent: Tuesday, March 21, 2006 1:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Problem with chan_iax.cimplimentationcausesbadaudio? On 21 Mar 2006, at 16:19, Adam Robins wrote: All switches and routers give highest priority to traffic on IAX2 port 4569. We use DSCB values over the IP-VPN to prioritize it as well. This did not change with the upgrade, as we can still see proper packet coding. The softphone is provided by our vendor Aheeva. It is the same IAX2 softphone they use in their own call centers. Funny thing is that they say that moving to Asterisk 1.2.4 tremendously IMPROVED their call quality with IAX2. Headsets are Plantronics H251N tops with DA60 USB adapters. All Desktops are at least 2.0 GHz P4 with 512MB RAM I don't suppose you have an ethereal packet capture from a bad call ??? Or a description of the 'badness'? I'm doing stuff in IAX2 at the moment and might be able to spot a problem. Tim. Tim Panton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Aterisk with Realtime
If you're using a mysql backend, it may be simpler to use res_mysql to get to it since it's designed specifically for use with mysql. Just change your configuration file extconfig.conf and change everything that says odbc over to mysql, and use the cdr_mysql plugin for the mysql connection. If you want to use odbc, make sure you can connect to the DSN with - isql DSN. If that connects, then you need to double check your res_odbc.conf (and cdr_odbc.conf, but I've never been able to get two odbc plugins working at the same time... we use cdr_pgsql for our cdr's) file to make sure it has the right info. Aaron On Tue, 21 Mar 2006, ram wrote: the DB in same server but iam using DSN to connect using ODBC is the not th right proces if not kindly recomend me the process i want to both SIP users / CDR to be from Mysql ram On 3/21/06, Aaron Daniel [EMAIL PROTECTED] wrote: Are you moving your db over to an odbc connection? Aaron On Tue, 21 Mar 2006, ram wrote: Hi i have installed unixODBC Drivers and created DSN but when i reload from * console asterisk terminating the services if i do again start the services and change something and reload again its hangs is that bug in 1.2.5 ram On 3/21/06, Aaron Daniel [EMAIL PROTECTED] wrote: Since you say you're using mysql as the backend, you need to change anything that says odbc to mysql so that the server knows where to find the db at. Also, you need to make sure the DB info is in res_mysql.conf. Aaron ram wrote: Hi thanks for the reply this what my extconfig sipusers = odbc,asterisk,2_Sip sippeers = odbc,asterisk,2_Sip extensions = odbc,asterisk,2_Extensions voicemail = odbc,asterisk,2_VMUsers voicemail_messages = odbc,asterisk,2_VM waht is wrong with this ? ram On 3/20/06, *Aaron Daniel* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: That, and make sure you've got extconfig set to use mysql for it's sippusers and sippeers and not odbc. Aaron Patrick wrote: On Mon, 2006-03-20 at 23:14 +0530, ram wrote: Hi iam working with asterisk with mysql Realtime when i have confgured and run the asterisk iam getting the following error i dig all the places for help could not find the results could some one help me what is wrong iam using 1.2.5 on FC4 Mar 20 23:04:52 NOTICE[2054] cdr.c: CDR simple logging enabled. Mar 20 23:04:52 NOTICE[2054] indications.c: Removed default indication country 'us' Mar 20 23:04:52 WARNING[2054] pbx_config.c: The use of '_.' for an extension is strongly discouraged and can have unexpected b ehavior. Please use '_X.' instead at line 3 Read the message and do as it suggests: in your dialplan replace all _. with _X. Patrick ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Realtime SIP Persistency
I've been using realtime for sip users information.
I noticed that when you are doing this, if you do a 'reload' or restart
asterisk, the information in a 'sip show peers' goes away. When I do this, MWI
stops working. I always though MWI used the astdb file ('database show') to
determine where to send MWI but it must be using 'sip show peers' because when
this is cleared, it stops working.
When I stop using realtime and instead provision users in sip.conf, a reload or
restart DOES NOT clear 'sip show peers'. It must be populating this list from
the astdb file in that case.
I'm going to scoot over to bugs.digium.com and report this as a bug, because
this is a real show stopper, and completely nullifies Realtime's use for us.
Doug
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Re: [Asterisk-Users] ISDN Protocol Unknom Error with Junghanns OctoBRI
On 3/21/06, Dave Cotton [EMAIL PROTECTED] wrote: span=1,1,3,ccs,ami I was going mad with a Quadbri until I changed ccs,ami to ccs,hdb3 and it's been running 6 months now. Dave, nice to read on this, can you explain what was going wrong when you used ccs,ami? And how did you find out about placing hdb3 there? As a quadbri user, I'm curious about this :) cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk with TOPEX GSM Gateway
Hi, I have 2 asterisk boxes connected both through internet with 8Mbit via IAX trunking. In asterisk A I have one TE410P card with one E1 active and I receive calls from PSTN and send them to asterisk B. In asterisk B I have other TE410P and one port is connected to one TOPEX GSM Gateway for outgoing calls to GSM network. Anyone using TOPEX with Asterisk connected with E1 interface? I have problems about quality of calls and ASR indicator is really low. I tested to use other IP gateways and it was fine. Waiting comments Tnks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP prepaid billing
Voipers Portugal wrote: Hi, I am using the following architecture: SER (SIP server) -- Asterisk -- PSTN Gateway And I would like to implement a prepaid billing solution that could be controlled by asterisk. Can anyone give me a direction? I use this setup with Asterisk2Billing (http://www.asterisk2billing.org/) and find it works well. You will find others at http://www.voip-info.org/, along with a wealth of other useful Asterisk information. Hope this helps. -- -Barry Flanagan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with intermittent one-way audio
Peter Fern wrote: I've had the same problem with all boxen running the same version. We ditched IAX2 for SIP and it has been working fine since. Well, upgrading my remote site to 1.2.5 appears to have fixed my issues. -Barry Doug Lytle wrote: Barry Flanagan wrote: Hi, I have a 1.2.4 asterisk box at a remote location, which is using IAX2 to connect to a 1.2.5 box for PSTN. There are 15 users on the remote server, all connecting via SIP softphones. For some reason, there is an increasing number of calls where the callee does not get any audio although the caller can hear them perfectly. I've had this problem in the past, when not running the same version of Asterisk on both ends of the trunk. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -Barry Flanagan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to make groups of extensions ???
Andrew Kohlsmith wrote: These are *all* incoming calls as far as Asterisk is concerned. You get dumped into a specific part of the dialplan (the context specified) and you tell Asterisk what they can dial. Internal extensions, external peers, Zap channels or even applications... the second half of all of this is the outgoing part, when Asterisk Dial()s. Do I not remember reading on this or the -dev list--I think more than once--that the recent reworkings (and future direction) of the SIP channel driver is to eliminate the notions of user/peer/friend for SIP, and have *all* endpoints be done as peers? Maybe I misunderstood a couple of previous threads, but I thought that for some time now (even though perhaps the previous configuration options regarding SIP users are still supported) that we have been asked not to maintain that distinction. I would welcome clarifying commentary from someone who is clued in on that matter. Not to say you're not, Andrew, because I might be misinformed. Thx. B. -A. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime SIP Persistency
Douglas Garstang wrote:
I've been using realtime for sip users information.
I noticed that when you are doing this, if you do a 'reload' or restart
asterisk, the information in a 'sip show peers' goes away. When I do this,
MWI stops working. I always though MWI used the astdb file ('database show')
to determine where to send MWI but it must be using 'sip show peers' because
when this is cleared, it stops working.
When I stop using realtime and instead provision users in sip.conf, a reload
or restart DOES NOT clear 'sip show peers'. It must be populating this list
from the astdb file in that case.
I'm going to scoot over to bugs.digium.com and report this as a bug, because
this is a real show stopper, and completely nullifies Realtime's use for us.
Doug,
I think this is documented behaviour. With realtime the peers do not
show up under sip show peers, and MWI does not happen. I think though if
you use rtcachefriends=yes in your [general] section of sip.conf that it
will work as you desire.
Hope this helps.
--
-Barry Flanagan
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Re: [Asterisk-Users] Realtime SIP Persistency
Try googling the archives using the keywords rtcachefriends mwi. You should find more info about this. regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to make groups of extensions ???
Andrew Kohlsmith wrote: On Tuesday 21 March 2006 12:25, Aaron Daniel wrote: Yeah, I agree with Chuck. User's on our system are put into various contexts depending on who they can call... local, long distance, or internal only. And *all* of those people are placing calls *in* to asterisk to get into those contexts. :-) When you pick up a telephone wired into an FXS port; asterisk sees an incoming request for dialtone. When you pick up your SIP phone and dial; it must match a friend or user entry or you'll never get in. When your IAX softphone client makes a call, again, it must match a friend or user entry. These are *all* incoming calls as far as Asterisk is concerned. You get dumped into a specific part of the dialplan (the context specified) and you tell Asterisk what they can dial. Internal extensions, external peers, Zap channels or even applications... the second half of all of this is the outgoing part, when Asterisk Dial()s. I think if you read the OP's original six posts, he's trying to address both incoming and outgoing calls. Eg., wants to reserver zap/4 for the boss for both. Then he had two other groups also that he wanted to only allow outgoing on selected zap channels. Not that difficult to address both. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoIP prepaid billing
Is it possible to use a2billing for everything except the dial out, as I want to use a2billing, to auth users and log time but I want to added custom IVR menus after users log in, like custom speed dial numbers. A2billing allows you to dial out no problem, but how do I get it to drop back to the main IVR and still monitor outgoing time? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Barry Flanagan Sent: Tuesday, March 21, 2006 2:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] VoIP prepaid billing Voipers Portugal wrote: Hi, I am using the following architecture: SER (SIP server) -- Asterisk -- PSTN Gateway And I would like to implement a prepaid billing solution that could be controlled by asterisk. Can anyone give me a direction? I use this setup with Asterisk2Billing (http://www.asterisk2billing.org/) and find it works well. You will find others at http://www.voip-info.org/, along with a wealth of other useful Asterisk information. Hope this helps. -- -Barry Flanagan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to make groups of extensions ???
Brian Capouch wrote: Andrew Kohlsmith wrote: These are *all* incoming calls as far as Asterisk is concerned. You get dumped into a specific part of the dialplan (the context specified) and you tell Asterisk what they can dial. Internal extensions, external peers, Zap channels or even applications... the second half of all of this is the outgoing part, when Asterisk Dial()s. Do I not remember reading on this or the -dev list--I think more than once--that the recent reworkings (and future direction) of the SIP channel driver is to eliminate the notions of user/peer/friend for SIP, and have *all* endpoints be done as peers? Maybe I misunderstood a couple of previous threads, but I thought that for some time now (even though perhaps the previous configuration options regarding SIP users are still supported) that we have been asked not to maintain that distinction. I would welcome clarifying commentary from someone who is clued in on that matter. Not to say you're not, Andrew, because I might be misinformed. I think you're right, at least that portion of what Olle posted on the topic. If I recall correctly, he was essentially suggesting matching on certain parameters (eg, IP address, username, etc) and doing away with the peer, user, friend terminology. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime SIP Persistency
David Thomas wrote: Try googling the archives using the keywords rtcachefriends mwi. You should find more info about this. Google doesn't work anymore; the subjects are listed just fine, but clicking on one leads to page not found. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP prepaid billing
Was it difficult to install and make it work? I need to make it work in a week or so, do you think it's possible? Did you manage to work with SER already? Because i don't see how can I distinguish the users because all of them come from SER, and don't Register directly into Asterisk. Jose SimoesOn 3/21/06, Barry Flanagan [EMAIL PROTECTED] wrote: Voipers Portugal wrote: Hi, I am using the following architecture: SER (SIP server) -- Asterisk -- PSTN Gateway And I would like to implement a prepaid billing solution that could be controlled by asterisk. Can anyone give me a direction?I use this setup with Asterisk2Billing(http://www.asterisk2billing.org/) and find it works well. You will find others at http://www.voip-info.org/, along with a wealthof other useful Asterisk information.Hope this helps.---Barry Flanagan ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Realtime SIP Persistency
I do have rtcachefriends=yes in sip.conf, and my astdb file is full of sip
contacts.
That's not the problem.
-Original Message-
From: Barry Flanagan [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 21, 2006 12:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Realtime SIP Persistency
Douglas Garstang wrote:
I've been using realtime for sip users information.
I noticed that when you are doing this, if you do a
'reload' or restart asterisk, the information in a 'sip show
peers' goes away. When I do this, MWI stops working. I always
though MWI used the astdb file ('database show') to determine
where to send MWI but it must be using 'sip show peers'
because when this is cleared, it stops working.
When I stop using realtime and instead provision users in
sip.conf, a reload or restart DOES NOT clear 'sip show
peers'. It must be populating this list from the astdb file
in that case.
I'm going to scoot over to bugs.digium.com and report this
as a bug, because this is a real show stopper, and completely
nullifies Realtime's use for us.
Doug,
I think this is documented behaviour. With realtime the peers do not
show up under sip show peers, and MWI does not happen. I
think though if
you use rtcachefriends=yes in your [general] section of
sip.conf that it
will work as you desire.
Hope this helps.
--
-Barry Flanagan
___
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___
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RE: [Asterisk-Users] Realtime SIP Persistency
I have rtcachefriends=yes in sip.conf. It is caching friends because as I said in my post, astdb has all the contacts, ie they're cached. It's the behaviour of 'sip show peers' that's not working. -Original Message- From: David Thomas [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 21, 2006 12:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Realtime SIP Persistency Try googling the archives using the keywords rtcachefriends mwi. You should find more info about this. regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with chan_iax.cimplimentationcausesbadaudio?
On 21 Mar 2006, at 18:35, Adam Robins wrote: Thanks for the offer. We deleted all of our Ethereal traces once we switched to SIP. On a bad call call there were tens of thousands of checksum errors and packets out of sequence. This occurred both with and without IAX2 trunking and trunktimestamps. Checksum errors- That is interesting - IAX doesn't have checksums, udp can though. Sounds like some aspect of IAX was interacting (in a bad way) with your VPN. The out of sequence stuff could definitely be the IAX update. I had to add code to cope with blocks of 3 packets arriving in the wrong order (over a WAN) when I moved to 1.2 Complaints of poor quality were from both the agent and customer sides. Mostly cutting in and out - typical of dropped packets. Yeah, that is consistent with the checksum errors. I'd be very interested if anyone else has packet traces that show this sort of problem, we are looking to deploy on IAX, and a bit of warning would be great. T. Tim Panton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sound dies
Hi guys, I'm using SIP phones with Asterisk 1.2 and going fine, most of the time. However, when the duration of a call is longer than 20minutes I often stop hearing the other party, but that one keeps most of the time hearing me. Does any of you know of this or similar problems? Thing is that I'm connected to the Asterisk over VPN tunnel, but how sensitive is the SIP protocol to a short glitches in connection? BR. Arnar Gestsson -- Arnar Gestsson [EMAIL PROTECTED] Trackwell Software ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Native MOH - Convert mp3 to ulaw
I don't know for sure about the formats, but I'd try sox. I'm pretty sure pcm/ulaw is built in... Bob McDowell -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Tuesday, March 21, 2006 11:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Native MOH - Convert mp3 to ulaw I'd like to use native moh instead of with mpg123... for some reason the processes never bloody die. For native moh to not spawn an external player, I'd need to convert the default supplied moh sound files in /var/lib/asterisk/mohmp3 to ulaw and g729 format. Anyone know of a free, easy way to convert them? Thanks, Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
