[Asterisk-Users] ipvolution TDM cards - vaporware?
Has anyone on this list gotten hold of these cards? It's been 2 months since their official ship date. Even the website www.ipvolution.com is in wee-wee land. /leo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk security problem: authorized SIP users can fake any callerid!
On Fri, 11 Mar 2005 14:41:37 -0500, C F [EMAIL PROTECTED] wrote: Welcome to SIP, this is how SIP works, thats why ppl use IAX. It is a combination of chan_sip and the particular sip.conf actually. Sane SIP servers will challenge all INVITEs, and apply user identification from the user database, not what the UA choose to supply. But if you configre Asterisk to accept anything from anyone, well you should expect this. Tom ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Parked Call
Try using dtmfmode=rfc2833 in your sip.conf. It should work... Hope, this could help. Guido Hecken I have a question, I am unclear on how to park a call. I know that you are supposed to be able to press # and then transfer the call to extension 700. However, * doesn't seem to be graping the dtmf. I am using dtmfmode=inband. Asterisk is in the media path as well. Thanks in advance Justin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Simultaneous call to both phones in PAP2-NA
J Thomas wrote: We have given a few PAP2-NA to our business customers with both phone ports configured through the same SIP server. We cannot call them both at the same time. Surprisingly, we can call both the phones one at a time fine. Is there something we are missing in the configuration? Any help in resolving this will be greatly appreciated. What codec are you using? Keep in mind that the PAP2 (and the Sipura SPA-2000 it is based on) can only do _one_ low-bandwidth connection at a time (G.729 or similar). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP monitor thread is hanged up on a uClinux embeded linux system
I met a strange SIP problem recently. In an ordinary procedure, when asterisk loads sip module, a series of functions are called sequentially: load_module()-restart_monitor()-ast_pthread_create()-pthread_create()-do_monitor() However in my system, pthread_create() failed to create a child thread to execute do_monitor(), (though pthread_create() returns a successful signal to asterisk.) Therefore my sip monitor can't monitor the incoming network flow and it results that no SIP packet will be caught. Could anyone give me some hints to explain why the thread is not created properly or why the thread is created but it doesn't execute do_monitor at all. Thanks a lot! Best Regards, Liang ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 Sphone for PocketPC
Does anobody know an IAX2 software phone for PocketPC? Regards, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X-Lite and * SIP Problem
Hi, I am playing around with SIP extensions on my local lan using X-Lite but I am having a bit of difficulty, I have set up X-Lite and my sip.conf accordingly, but when I start it I get the following message: Login failed! Contact Network Admin I am still able to dial local extensions on my * with x-lite even though it is in this state, although trying to dial my sip extension from a real extension results in a busy tone. Every 30 seconds or so, my asterisk console shows the following message: Mar 12 09:48:00 NOTICE[802]: chan sip.c 8448 handle request: Registration from 'richard sip:[EMAIL PROTECTED]' failed for '192.168.0.100' My sip.conf is as follows: [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes [200] type=friend username=richard secret=password host=dynamic reinvite=no canreinvite=no dissallow=all context=sip allow=gsm And my X-lite Default SIP Proxy config is as follows: Enabled: Yes Display name: richard Username: richard Authorisation User: richard Password: password Domain/Realm: 192.168.0.102 (my asterisk server's IP) Sip Proxy: 192.168.0.102 rest left as default Can anyone tell me what I'm doing wrong? This is all through a local lan, no nat or anything. Thanks, Richard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Location of Voice e-mail Code???
Hi, Where can I find the code that performs the voice e-mail function (that is, the code that reads the contents of voicemail.conf and then performs the necessary action)? I am using [EMAIL PROTECTED] 0.6. Thanks in advance! -- Rgds, Julius. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [Asterisk-Users] GotoIf problem
-Oprindelig meddelelse-
Fra: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] På vegne af kurt x
Sendt: 9. marts 2005 20:57
Til: Chris Wade
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: Re: [Asterisk-Users] GotoIf problem
I,ve gotten the GotoIf statement working now. I hard coded the value
10 in place of the ${DIGITS} varible. Worked like a charm.
Thanks to everyone who helped.
Kurt
Hi Kurt,
You are writing the ${DIGITS} variable wrong, you are missing a {
eg.: you are writing $DIGITS} and it should be ${DIGITS}
Thorben
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Re: [Asterisk-Users] ipvolution TDM cards - vaporware?
Leo Ann Boon wrote: Has anyone on this list gotten hold of these cards? It's been 2 months since their official ship date. Even the website www.ipvolution.com is in wee-wee land. It has been down for several weeks. The cards are still shown on www.atacomm.com. I don't know whether that is a positive sign, or if things just haven't been updated. Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Location of Voice e-mail Code???
On Sat, 12 Mar 2005 13:03:00 +0300 (EAT), Julius Kidubuka [EMAIL PROTECTED] wrote: Hi, Where can I find the code that performs the voice e-mail function (that is, the code that reads the contents of voicemail.conf and then performs the necessary action)? I am using [EMAIL PROTECTED] 0.6. Thanks in advance! cd /usr/src/asterisk grep -r voicemail.conf * should give you a clue or two. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail to UK mobile
What issues/options are there when forwarding voicemail to uk mobile voicemail? ta Rafal Kaniewski -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.7.2 - Release Date: 11/03/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Droping calls
I have no idea. I live in the USA so I don't normally need busydetect. Anton Krall wrote: Why does busydetect actually drop calls while stile talking? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Viernes, 11 de Marzo de 2005 03:58 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Droping calls Anton Krall wrote: Guys, this is weird.. Today I started having some problems with calls been dropped. Im suing X100p cards (clones) and I have this setting on my zatala fle: [channels] [snip] busydetect=yes busycount=4 Can the echotraining be messing things? Do I need to enable callprogress or something? What do guys think? callprogress and busydetect should both be renamed to randomlydisconnectmycalls because that is what they do. You can REDUCE the number of random disconnects by increasing the value of busycount. There's not anything you can do about disconnects caused by callprogress. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VegaStream 50 BRI
Hi there, Can anyone help me with a problem I have setting up my VEGA 50 BRI gateway with Asterisk? I have been successful making outgoung calls, but have been unable to get the Vega to register with Asterisk. Would anyone have a sample section of Sip.conf to help me? Does Asterisk currently support SIP2 as I think the VEGA does. Many thanks, Colin Holman. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.7.2 - Release Date: 11/03/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Location of Voice e-mail Code???
On Sat, Mar 12, 2005 at 01:03:00PM +0300, Julius Kidubuka wrote: Hi, Where can I find the code that performs the voice e-mail function (that is, the code that reads the contents of voicemail.conf and then performs the necessary action)? I am using [EMAIL PROTECTED] 0.6. The mail is delivered by piping it to a sendmail program (by default /usr/sbin/sendmail). /usr/sbin/sendmail does not have to be sendmail. Postfix and Exim provide a sendmail-compatible interface along with a host of more minimal programs such as ssmtp and nullmailer. With sendmail and similar (Exim and Postfix) the aliases (normally /etc/aliases) file is a useful place to set up forwarding. e.g: suppose you want to keep your voicemail.conf as simple as possible: [default] #vmbox=pass,name,recipients 200=200,,[EMAIL PROTECTED] 201=201,,[EMAIL PROTECTED] 202=202,,[EMAIL PROTECTED] 202=202,,[EMAIL PROTECTED] 203=203,,[EMAIL PROTECTED] 204=204,,[EMAIL PROTECTED] to your aliases file you could then add: 200: john 201: [EMAIL PROTECTED] 202: david,|/usr/local/bin/send_sms_to_david Note that I have ommited the names, but those names are actually also used for things other than voicemail. -- Tzafrir Cohen | New signature for new address and | VIM is http://tzafrir.org.il | new homepage | a Mutt's [EMAIL PROTECTED] || best ICQ# 16849755 | Space reserved for other protocols | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Print-to-Fax client
Florian Overkamp wrote: Hi, -Original Message- You should be able to download one (for WIndows and possibly Mac) from efax or j2.com I think. http://www.efax.com/en/efax/twa/page/download?rqcp=2 http://www.j2.com/jconnect/twa/page/download You might be able to do that, but take a good look at the license agreement on the driver - you might not be allowed to use the software fully without having a subscription to their services. Florian Just as an extra datapoint, I have been experimenting with the builltin fax facilities in OSX. Basically there is a cups printer called Internal_Modem which accepts postscript and sends faxes out of the external modem. I had the external modem plugged into a sipura 2000 which connected over ulaw to asterisk which then calls out over PRI to the far fax. Initial experiments seem encouraging . The really cool thing is that the CUPS printer is a network service so all the systems on my net can now send faxes. The othernice thing is that the processing of postscript to tiff/fax is done off the asterisk box with a full set of genuine postscript fonts. (With the imac mini - it could almost be described as cost effective to buy one just as a fax gateway). Tim. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ATA 186 Codec Question.
I have seen the list of codecs for the ATA 186's but not sure if it was 100% or not. I want to know really is it possible to run GSM or ilbc on them or is a G729 lic the only way to get a low bandwidth codec? This is the list of codecs that I have seen. RxCodec and TxCodecConfigure the codec ID. * G.723.1Codec ID 0 * G.711aCodec ID 1 * G.711ucodec ID 2 * G.729acodec ID 3 Thanks David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Best DB
[EMAIL PROTECTED] wrote: I know this is a bit off topic but we are using Asterisk :) Since this list is full of tech gurus w/ all different sorts of backgrounds, I thought I would get the best opinions here. We have several different switches and other telecom equipment at our facilities which all have their own proprietary cdr platforms, which are rather limited. The company I work for is looking to develop their own in-house billing system that would combine cdr from all platforms and bring it into one big db, so we can do whatever we like w/ the data...billing, invoices, reports, asr...etc... So my question is this What's the most stable, fastest reliable database for this project? Call volume is about 8 to 10 million minutes per month, and we want to have 12 months of cdr available at any given time, anything older can be archived on tape. So what's the best db...oracle, ms sql, informix, mysql or something else? At the risk of sounding like a closed source fan (I'm not) I do think you should at least consider Oracle for this job. I built a system a few years ago which takes a constant stream of entries from a number (100) of remote systems analizes them and generates reports (see http://www.westpoint.ltd.uk/example-reports/reports/index.htm) We use Oracle for it, and it has been great. Also they have improved the weakest points: 1) pricing - It is now _much_ cheaper than it was 2) Install - I had a couple of oracle newbies install it in a couple of hours, that was never possible in the old days. Once you have it there are _stacks_ of neat features and a really solid performance. I am especialy fond of the ability to put java into triggers (we send SNMP traps to ops console when specific error conditions occur on inserts) and the whole oracle Text and XML integration has saved me _months_ of development time on various project. My view is that if you are going to spend significant development time/money on a big database project, you shouldn't rule oracle out 'cos of a 2k fee. Tim. What server specs would be ideal for this type of setup? TIA, Jon --- [This E-mail scanned for viruses by Declude Virus] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple IAX Phones Behind NAT
Ok, I've seen this question go unanswered on the mailing list, and I assume it's because no one had the heart to break the bad news to the guy asking, but be honest with me, I can take it. At this time it's flat impossible to have multiple IAX phones behind a NAT without using an * gateway because there's no way to have a client listen on a port besides 4569. Is my only option to learn about SIP and attempt to forward that through my NAT? Multiple iax phones behind a nat box is known to function correctly. However, some nat boxes do not properly handle this. The easiest way to analyze the problem is to use a packet sniffer (eg, ethereal) on the outside of the nat box to see what it's doing to you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] EADS6550 and asterisk - echo on PSTN call
would like to know if some of you have tested asterisk connected to an EADS 6550 analogique PBX (also know as Nexpan50). Our set up is a Dell Optiplex with 1 TDM400 4 FXO, 1 TDM400 4 FXS, (no other card, each of them have their own IRQ) all ports connected to the EADS. We have GS ATA286 as EP. Calling from ATA to ATA give a perfect sound. Calling to PSTN numbers or reverse side, give echo. We can do what we want in zconfig.h (STEVE2, MARK2, MMX, AGGRESSIV_SUPPRESSOR, NOECHOCAN_DISABLE) or zapata.conf (tx gain=-10.0, echocancel=32 ... 256,), test with differents POTS phone, it change nothing. We even didn't notice changes between our various changes in those files (and yes modules where unloaded between each test). Always the same echo. So know we start to doubt that this echo problem is asterisk related but perhaps more to the PBX. That's why we ask if some of you have/had similar setup with this PBX and if there is a solution. You didn't mention what country your in; if you outside the US, be sure to config the TDM-fxo card for your country (eg, line impedance). You mention echocancel=32, etc, did you try echotraining=800? For my TDM-fxo in the US, using the following on each channel works fine: echotraining=800 echocancel=yes echocancelwhenbridged=yes rxgain=5.0 txgain=0.0 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM04B lock up
I have a strange situation. Once in a while (non-deterministic) the 2 TDM04B cards lock up at the same time and stop processing incoming and outgoing calls even though * shows that it is trying to communicate to ZAP channels (at least on the outgoing). The only cure is to reboot the system when it happens. It makes me very apprehensive of the system Has anyone seen this problem. Could this be something to do with the IRQ sharing. Here is the output of lspci -v. I see that one of the cards shares IRQ # with VGA controller and the other one with ICH4 IDE. Any help would be appreciated. The TDM04b card is known to do that and its been reported to digium by several users. Seems to lock up about once every week or two, and varies from one system to another. The digium folks aren't saying anything as yet about what needs to be fixed. But, best open a trouble ticket with them for the records. There is a very high probability it is a hardware design issue. I've added a capacitor to my card under the assumption that noisy on-card logic is distrubing the chipsets. So far I'm not had a lockup, but it's a little to early to say that actually fixed it. irq sharing has been known to be an issue in some cases, however its generally over-blown on this list. If you want to eliminate that as an issue, either move you cards around to different slots or take a look in your bios config to disable those irq's on in use (or both). I'd guess the probability of an irq issue causing your problems is in the neighborhood of about a 20% chance. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA 186 Codec Question.
David Uzzell wrote: I have seen the list of codecs for the ATA 186's but not sure if it was 100% or not. I want to know really is it possible to run GSM or ilbc on them or is a G729 lic the only way to get a low bandwidth codec? This is the list of codecs that I have seen. RxCodec and TxCodecConfigure the codec ID. * G.723.1Codec ID 0 * G.711aCodec ID 1 * G.711ucodec ID 2 * G.729acodec ID 3 The Cisco ATA-186 only supports those codecs. Recent firmware also supports G726. You are not going to run GSM or iLBC codecs with the Cisco ATA-186. --Eric -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Signaling on PRI channels
Hi, We did an interconnection with our carrier few days ago. But, I noticed that there was a signaling problem on our trnuk. In fact, Asterisk indicates that the call is answered when we received ALTERTING message from our carrier. This is PRI debug logs : -- Executing Dial(IAX2/[EMAIL PROTECTED]/5, ZAP/g1/0130450836) in new stack -- Making new call for cr 32784 Protocol Discriminator: Q.931 (8) len=33 Call Ref: len= 2 (reference 16/0x10) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a9 83 82] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 2 ] [6c 02 00 c3] Calling Number (len= 4) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) Presentation: Number not available (67) '' ] [70 0b 80 30 31 33 30 34 35 30 38 33 36] Called Number (len=13) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '0130450836' ] [a1] Sending Complete (len= 1) -- Called g1/0130450836 Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 16/0x10) (Terminator) Message type: CALL PROCEEDING (2) [18 03 a9 83 82] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 2 ] -- Processing IE 24 (cs0, Channel Identification) Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 16/0x10) (Terminator) Message type: PROGRESS (3) [08 02 87 f2] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: International network (7) Ext: 1 Cause: Unknown (114), class = Interworking (7) ] [1e 02 82 81] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Progress Description: Call is not end-to-end ISDN; further call progress information may be available inband. (1) ] -- Processing IE 8 (cs0, Cause) -- Processing IE 30 (cs0, Progress Indicator) Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 16/0x10) (Terminator) Message type: ALERTING (1) -- Zap/2-1 is ringing -- Zap/2-1 answered IAX2/[EMAIL PROTECTED]/5 Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 16/0x10) (Terminator) Message type: CONNECT (7) Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 16/0x10) (Originator) Message type: CONNECT ACKNOWLEDGE (15) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 16/0x10) (Terminator) Message type: DISCONNECT (69) [08 02 87 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: International network (7) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Processing IE 8 (cs0, Cause) -- Channel 0/2, span 1 got hangup NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 16/0x10) (Originator) Message type: RELEASE (77) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Hungup 'Zap/2-1' == Spawn extension (PlugAndTel_default, 0130450836, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]/5' -- Hungup 'IAX2/[EMAIL PROTECTED]/5' This is the zapata.conf file : [channels] context=default switchtype=euroisdn pridialplan=unknown usecallerid=yes hidecallerid=yes callwaiting=yes usecallingpres=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 immediate=no callprogress=yes musiconhold=default ; Span 1 switchtype = euroisdn signalling = pri_cpe group = 1 context = default channel = 1-15 channel = 17-31 ; Span 2 switchtype = euroisdn signalling = pri_cpe group = 2 context = default channel = 32-46 channel = 48-62 ; Span 3 switchtype = euroisdn signalling = pri_cpe group = 3 context = default channel = 63-77 channel = 79-93 ; Span 4 switchtype = euroisdn signalling = pri_cpe group = 4 context = default channel = 94-108 channel = 110-124 The problem appears on span 1. This results with a bad billing. Thansk for your help. Laurent. ___ Asterisk-Users mailing list
RE: [Asterisk-Users] Asterisk on MS Virtual Server
I think you missed out a *NOT* below... In short, you *CANNOT* install or otherwise use any hardware cards, like Zaptel, with Asterisk when running on CoLinux and generally, I'll advise you to not use Astwind for anything other then playing. It's a nice toy, but that is all. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gilad Ben-Yossef Sent: 02 March 2005 14:59 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk on MS Virtual Server Turgut Abacioglu wrote: Hello I downloaded Astwind and get working the network (means can access to Internet through MS Windows). DEbian and Asterisk files are updated from Internet. But When I make install in Zaptel (it was my first make) I got many errors. Acoording to one manual this happens when we do not have modeversion .h kernel header file (according to it, it should reside in /usr/src/linux) which in /usr/src/linux, a make menuconfig will create it. BuT I do not have the linux dir (in /usr/src) and kernel source files thus modversion.h file. In addition I do not know how to download kernel files to linux directory (I tried apt-get but I could not format properly the /etc/spt/source.list file) Could you help. Am I in the correct path? No, you are not. Zaptel is a driver to hardware cards. CoLinux (on which Astwind is based) is a virtual Linux running as a Windows task. Virtual here means - no hardware. In short, you can install or otherwise use any hardware cards, like Zaptel, with Asterisk when running on CoLinux and generally, I'll advise you to not use Astwind for anything other then playing. It's a nice toy, but that is all. Gilad -- Gilad Ben-Yossef [EMAIL PROTECTED] Codefidence. A name you can trust(tm) Web: http://codefidence.com | SIP: [EMAIL PROTECTED] Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643 I am Jack's Overwritten Stack Pointer -- Hackers Club, the movie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No ringback over IAX - LiveVoip
Grett. This should be loads of fun then... 8( I have noticed what I can only describe a negative undertone with several VoIP poviders. Not an easy customer? We don't want you. Things like that. The LiveVoIP website is in fact like that. There are several places on the site that just flat out say are you customer type x?, we don't want you then. Not my way of doing business but to each their own. I guess as long as they service my account and provide a good voip connection it won't mean much to me. There are more then a few folks using * that try various itsp services without a clue as to how to make things work, and/or with unrealistic expectations. I happen to like their no-nonsense approach on their web site of getting your attention. It sort of resembles some of the postings on this list relative to 'did you try to look for doc'. Overall, I'd give livevoip high marks in both quality and service. Let's hope they can keep it up. :) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7970
Hi, Is there anyone with some uptodate info on the Cisco 7970 and Asterisk skinny / SCCP? I know chan_sccp doesn't support the 7970. Is it true the basics should work with skinny? All info is welcome. Steven ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Droping calls
Guys, this is weird.. Today I started having some problems with calls been dropped. Im suing X100p cards (clones) and I have this setting on my zatala fle: busydetect=yes busycount=4 Try changing busycount to 6 or 8, stop asterisk, and restart. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Fwd: RE: [Asterisk-Users] Re: TE410P card in an HP-Compaq DL380 G4 server]
How did you go? On Tue, 8 Mar 2005 11:28:59 +1030, Peter Childs [EMAIL PROTECTED] wrote: Digium shipped me a replacement card, but they sent the wrong one, so they fedex'd another and its just arrived. Should be testing in the next two days (the box is in another state...) The last I heard from Eric Bishop (on the 1st march) was that he had received an updated card from digium, but it didn't function in his DL380... I can let you know the outcome of the test if you'd like. Cheers, Peter -Original Message- From: Mark F. Vickers [mailto:[EMAIL PROTECTED] Sent: Tuesday, 8 March 2005 11:20 AM To: [EMAIL PROTECTED]; asterisk-users@lists.digium.com Subject: [Fwd: RE: [Asterisk-Users] Re: TE410P card in an HP-Compaq DL380 G4 server] Was there any resolution on this I also have a TE410P in an box with an Intel E7501 chipset? -Vickers Original Message Subject: RE: [Asterisk-Users] Re: TE410P card in an HP-Compaq DL380 G4 server Date: Tue, 8 Feb 2005 11:13:24 +1030 From: Peter Childs [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com CC: [EMAIL PROTECTED] RMA your non-functional card and get one with a new firmware they are trying that fixes the issues with the Intel E75xx chipsets. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Tony Mountifield Sent: Monday, 7 February 2005 6:53 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: TE410P card in an HP-Compaq DL380 G4 server In article [EMAIL PROTECTED], Peter Childs [EMAIL PROTECTED] wrote: Contact Digium Support. They have been very helpful with this issue (mention your using the G4 server with the Intel E7520 Chipset..) So do they have a solution? What is it? Cheers Tony ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] EADS6550 and asterisk - echo on PSTN call
Rich Adamson a écrit : would like to know if some of you have tested asterisk connected to an EADS 6550 analogique PBX (also know as Nexpan50). Our set up is a Dell Optiplex with 1 TDM400 4 FXO, 1 TDM400 4 FXS, (no other card, each of them have their own IRQ) all ports connected to the EADS. We have GS ATA286 as EP. Calling from ATA to ATA give a perfect sound. Calling to PSTN numbers or reverse side, give echo. We can do what we want in zconfig.h (STEVE2, MARK2, MMX, AGGRESSIV_SUPPRESSOR, NOECHOCAN_DISABLE) or zapata.conf (tx gain=-10.0, echocancel=32 ... 256,), test with differents POTS phone, it change nothing. We even didn't notice changes between our various changes in those files (and yes modules where unloaded between each test). Always the same echo. So know we start to doubt that this echo problem is asterisk related but perhaps more to the PBX. That's why we ask if some of you have/had similar setup with this PBX and if there is a solution. You didn't mention what country your in; if you outside the US, be sure to config the TDM-fxo card for your country (eg, line impedance). France. You mention echocancel=32, etc, did you try echotraining=800? Yes. It create a second echo :-( For my TDM-fxo in the US, using the following on each channel works fine: echotraining=800 echocancel=yes echocancelwhenbridged=yes rxgain=5.0 txgain=0.0 Do you have this setup with the standard zconfig.h (MARK2)? Thanks for your reply. -- Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No ringback over IAX - LiveVoip
I saw some coverage of this in the list archive but no one seems to have posted a resolution. I am using [EMAIL PROTECTED] 0.06 and when I get a call from LiveVoip over IAX I dump it into my IVR. From there the call is routed to groups based upon input. However, there is no ringback indicated to the IAX caller. Perhaps they expect you to provide audioable progress information inband on the reverse channel? I.e. use the 'r' option on the dial command etc. That is the way some isdn lines etc work. Peter That is what everyone is bitching about. No matter whether you use the r option or not, you never get ringback through LiveVoIP. And they consistently point the finger at * rather that trying to solve the problem. If you use ethereal to inspect the iax packets in the above case, you see that asterisk is sending an IAX Type=Control packet with a Control subclass: Ringing (3) to the LiveVoip switch. LiveVoip is ignoring that particular control packet. I'd have to guess that LiveVoip wants ringback to occur in the audio stream, not as a iax control packet, and therefore is blaming asterisk. The r option within asterisk (in the above case) is doing exactly what Mark intended for asterisk-to-asterisk iax connections, which is different then LiveVoip expectations. So, who is wrong here, or is this just human translations of what is expected in a non-rfc communications environment? If asterisk is going to be modified to support LiveVoip expectations, then yet another Dial option would need to be implemented to force ringback to occur as an audio stream for iax only. Guess one could open a bug report for both LiveVoip and Asterisk, but not likely to be addressed any time soon since this is itsp dependent. Rich ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura 2100 and Asterisk one-way audio
Hello People, I have a Sipura SPA-2100 with default configuration and the last software upgrade, and a * from Debian Sarge with the simple configuration: [general] port = 5060 bindaddr = 0.0.0.0 [103] username=103 type=friend secret=qaz123wsx qualify=no port=5060 nat=yes host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=X 103 allow=all I have the normal codecs (Debian Installation). The Sipura was registered OK, I make a call, I listen but the other site cannot. I can call, with the Sipura, direct to other SIP Phone without problem. I change the Sipura SPA-2100 with a Sipura SPA-1000 in the same peer and it work fine. It is the same problem in other Sipura SPA-841. Do you have any clue about it? Thank in advance Charlie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wireless VoIP
we have a full line of WiFi phone that work great. Works at any hot spot also for more info contact me at [EMAIL PROTECTED] - Original Message - From: Sylvain COUTANT [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, March 11, 2005 12:42 PM Subject: [Asterisk-Users] Wireless VoIP Hi all, I have no experience with wireless VoIP. Do you have some quality wireless phones to suggest ? Thanks in advance. Sylvain COUTANT http://www.adviseo.net/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] EADS6550 and asterisk - echo on PSTN call
would like to know if some of you have tested asterisk connected to an EADS 6550 analogique PBX (also know as Nexpan50). Our set up is a Dell Optiplex with 1 TDM400 4 FXO, 1 TDM400 4 FXS, (no other card, each of them have their own IRQ) all ports connected to the EADS. We have GS ATA286 as EP. Calling from ATA to ATA give a perfect sound. Calling to PSTN numbers or reverse side, give echo. We can do what we want in zconfig.h (STEVE2, MARK2, MMX, AGGRESSIV_SUPPRESSOR, NOECHOCAN_DISABLE) or zapata.conf (tx gain=-10.0, echocancel=32 ... 256,), test with differents POTS phone, it change nothing. We even didn't notice changes between our various changes in those files (and yes modules where unloaded between each test). Always the same echo. So know we start to doubt that this echo problem is asterisk related but perhaps more to the PBX. That's why we ask if some of you have/had similar setup with this PBX and if there is a solution. You didn't mention what country your in; if you outside the US, be sure to config the TDM-fxo card for your country (eg, line impedance). France. You mention echocancel=32, etc, did you try echotraining=800? Yes. It create a second echo :-( For my TDM-fxo in the US, using the following on each channel works fine: echotraining=800 echocancel=yes echocancelwhenbridged=yes rxgain=5.0 txgain=0.0 Do you have this setup with the standard zconfig.h (MARK2)? I might have missed some of your earlier posts relative to this; just catching up on over 500 emails from this list. I've not had to configure a TDM for non-US support, but I know for an absolute fact (based on 20 years of detailed telephony engineering experience) that you have to config the TDM card for line impedance, etc, for your country. If you've not done that, start there. (Think that's an optional parameter when loading the drivers.) I update asterisk from cvs-head about every two weeks or so, and always stick with default values (including zconfig.h). So, yes I'm using the default echo cancellation, etc. There has not been very many changes associated the the zaptel source code and the TDM-fxo drivers. Certainly not necessary to use the latest cvs-head at all; anything from the last few months should work. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to create channel of type 'IAX2'
Hi all,
I'm a newbie and I have a configuration problem with Asterisk.
Seems that I'm not able to call an outbound number. I'm quite sure that it
is a configuration problem, but I'm not able to find out where is the
mistake, even reading several docs to www.voip-info.org.
I do not have a good knowledge of Asterisk, I'm not very familiar with its
configuration and I've a big confusion about it.
Any help will be appreciated.
When I try to call outside I got the following message:
*CLI -- Accepting AUTHENTICATED call from 192.168.0.55:
requested format = ilbc,
requested prefs = (),
actual format = gsm,
host prefs = (gsm|ilbc|speex),
priority = mine
-- Executing Dial(IAX2/[EMAIL PROTECTED]/2, IAX2/3479450772) in new stack
Mar 12 16:15:36 WARNING[3149]: chan_iax2.c:2341 create_addr: No such host:
3479450800
Mar 12 16:15:36 NOTICE[3149]: app_dial.c:911 dial_exec_full: Unable to
create channel of type 'IAX2' (cause 3)
== Everyone is busy/congested at this time (1:0/1/0)
Mar 12 16:15:47 WARNING[3149]: pbx.c:2028 ast_pbx_run: Timeout, but no rule
't' in context 'fullaccess'
-- Hungup 'IAX2/[EMAIL PROTECTED]/2'
My zapata.conf is:
[channels]
language=it
context=fullaccess
signalling=fxs_ks
usecallerid=yes
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
pickupgroup=1
immediate=yes
musiconhold=default channel = 1
channel=1
My iax.conf is:
[general]
bindaddr=0.0.0.0
context=noaccess
group=1
callgroup=1
pickupgroup=1
amaflags=default
bandwidth=low
disallow=all ; same as bandwidth=high
disallow=ulaw
disallow=alaw
allow=gsm
allow=iLBC
allow=Speex
jitterbuffer=yes
dropcount=2
maxjitterbuffer=500
maxexccessbuffer=400
tos=throughput
[guest]
type=user
context=iaxguest
callerid=Guest IAX User
[emi]
type=friend
username=emi
secret=none
auth=md5
host=dynamic
context=fullaccess
mailbox=101
callerid=Emi102
iax.conf 44L, 621C
and my extension.conf is:
[general]
static=yes
writeprotect=yes
[fullaccess]
include = parkedcalls
include = local
[local]
exten = _XX,1,Dial(IAX2/${EXTEN})
Where is the mistake?
Regards,
___
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Re: [Asterisk-Users] X-Lite and * SIP Problem
[200] type=friend username=richard change this to username=200 And my X-lite Default SIP Proxy config is as follows: Enabled: Yes Display name: richard Username: richard Change this to Username: 200 and this one Authorisation User: richard to Authorisation User: 200 and it should work hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime does not work yet, ...
Mark Matthew, I know how frustrating it may be, ... I can imagine your feelings, ... HOWEVER, with all respect, it does not help me to fix my problem! Can we come back to the subject, please? I apologies for the missing words for me in the Subject! I tried to follow (and may made some mistakes) all what was explained at the wiki. I have taken out one of my sip phones chapter and put this one as one record into the database. I fixed to add that it uses the right sock. (I do not understand why it was looking in /tmp instead reading /etc/my.ini to find it) Added in res_mysql.conf: dbport = 3306 dbsock = =/var/lib/mysql/mysql.sock I check if the record is in the mysql database: mysql select * from sip_buddies where name='621'; ++--+-+--+---+--+-+-+---+--+--++-+---+---+--+--+---+---+-++--+--+-+--+-+-+++---++--+---+--+-++++ | id | name | accountcode | amaflags | callgroup | callerid | canreinvite | context | defaultip | dtmfmode | fromuser | fromdomain | host| incominglimit | outgoinglimit | insecure | language | mailbox | md5secret | nat | permit | deny | mask | pickupgroup | port | qualify | restrictcid | rtptimeout | rtpholdtimeout | secret| type | username | allow | disallow | musiconhold | regseconds | ipaddr | cancallforward | ++--+-+--+---+--+-+-+---+--+--++-+---+---+--+--+---+---+-++--+--+-+--+-+-+++---++--+---+--+-++++ | 1 | 621 | NULL| NULL | NULL | Demo 621 | yes | inhouse | NULL | rfc2833 | NULL | NULL | dynamic | NULL | NULL | NULL | NULL | [EMAIL PROTECTED] | NULL | 1 | NULL | NULL | NULL | 1 | | 999 | NULL| NULL | NULL | Password | friend | 621 | ulaw;alaw | all | NULL| 0 || yes| ++--+-+--+---+--+-+-+---+--+--++-+---+---+--+--+---+---+-++--+--+-+--+-+-+++---++--+---+--+-++++ 1 row in set (0.00 sec) I restarted (not just reloaded) Asterisk and the first message Asterisk tells me is: The 'sipfriends' table is obsolete, update your config to use sipusers and sippeers, though they can point to the same table. extconfig.conf: sipfriends = mysql,astconf,sip_buddies sipusers = mysql,astconf,sip_buddies sippeers = mysql,astconf,sip_buddies To remark the line sipfriends stopped the first line message! sip show users and sip show peers does not show the phone, but that maybe is normal, since as I understand the database concept it will only asked if there should be a phone! (Correct me if I am wrong, please) To make a phone call from 601 to 621 gives me a person .. is unavailable: -- Executing Dial(SIP/601-9e81, SIP/621|60|Ttrm) in new stack Mar 12 22:49:41 NOTICE[25640]: app_dial.c:927 dial_exec_full: Unable to create channel of type 'SIP' (cause 3) == Everyone is busy/congested at this time (1:0/1/0) A call from 621 to 601, however, gives me a connection!!! -- Executing Dial(SIP/621-8cc5, SIP/601|60|tr) in new stack -- Called 601 -- SIP/601-c558 is ringing == Spawn extension (inhouse, 601, 1) exited non-zero on 'SIP/621-8cc5' sip show users and sip show peers still do not show anything. /var/log/astersisk/debug shows for the seconds of these events: Mar 12 22:49:41 DEBUG[25640]: Check for res for 601 Mar 12 22:49:41 DEBUG[25640]: build_route: Contact hop: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp Mar 12 22:49:41 DEBUG[25640]: Setting NAT on RTP to 524288 Mar 12 22:49:41 DEBUG[25640]: # Testing 61.220.121.190 with 192.168.0.0 Mar 12 22:49:41 DEBUG[25640]: Target address 61.220.121.190 is not local, substituting externip Mar 12 22:49:41 DEBUG[25640]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Found Mar 12 22:49:41 DEBUG[25640]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Found Mar 12 22:49:41 DEBUG[25640]: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_buddies WHERE name = '621' Mar 12 22:49:41 DEBUG[25640]: MySQL
Re: [Asterisk-Users] ASTCC or should I use something else for different rates, depending on the calling card?
Stephen Misel wrote: Ronald Wiplinger wrote: New developments in our business plan make a change necessary. We would like to offer different prices, depending on the user/calling card. How can we use that with ASTCC? or should we use something else? ASTCC allows for multiple brands and different prices for each. I believe the CVS version has my expiration and maintenance fee patches as well. I use the latest ASTCC version, but I do not see how you can use different cards / different prices. Can you explain that for me, please? As I see it: The routes depend on trunks, but the trunks do not care about the cards. bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] EADS6550 and asterisk - echo on PSTN call
Rich Adamson a écrit : would like to know if some of you have tested asterisk connected to an EADS 6550 analogique PBX (also know as Nexpan50). Our set up is a Dell Optiplex with 1 TDM400 4 FXO, 1 TDM400 4 FXS, (no other card, each of them have their own IRQ) all ports connected to the EADS. We have GS ATA286 as EP. Calling from ATA to ATA give a perfect sound. Calling to PSTN numbers or reverse side, give echo. We can do what we want in zconfig.h (STEVE2, MARK2, MMX, AGGRESSIV_SUPPRESSOR, NOECHOCAN_DISABLE) or zapata.conf (tx gain=-10.0, echocancel=32 ... 256,), test with differents POTS phone, it change nothing. We even didn't notice changes between our various changes in those files (and yes modules where unloaded between each test). Always the same echo. So know we start to doubt that this echo problem is asterisk related but perhaps more to the PBX. That's why we ask if some of you have/had similar setup with this PBX and if there is a solution. You didn't mention what country your in; if you outside the US, be sure to config the TDM-fxo card for your country (eg, line impedance). France. You mention echocancel=32, etc, did you try echotraining=800? Yes. It create a second echo :-( For my TDM-fxo in the US, using the following on each channel works fine: echotraining=800 echocancel=yes echocancelwhenbridged=yes rxgain=5.0 txgain=0.0 Do you have this setup with the standard zconfig.h (MARK2)? I might have missed some of your earlier posts relative to this; just catching up on over 500 emails from this list. I've not had to configure a TDM for non-US support, but I know for an absolute fact (based on 20 years of detailed telephony engineering experience) that you have to config the TDM card for line impedance, etc, for your country. If you've not done that, start there. (Think that's an optional parameter when loading the drivers.) I update asterisk from cvs-head about every two weeks or so, and always stick with default values (including zconfig.h). So, yes I'm using the default echo cancellation, etc. There has not been very many changes associated the the zaptel source code and the TDM-fxo drivers. Certainly not necessary to use the latest cvs-head at all; anything from the last few months should work. My /etc/zaptel.conf is adapted to country: loadzone=fr defaultzone=fr Asterisk stable 1.0.5. If you're telling that I have to pass parameters to module when loading, I checked with modinfo wctdm (at office I have head version) and options I have are those: [EMAIL PROTECTED] asterisk]# /sbin/modinfo -p wctdm debug int loopcurrent int robust int _opermode int opermode string timingonly int lowpower int boostringer int fxshonormode int battdebounce int battthresh int alawoverride int Pardon my ignorance but no one of them remaind me to impedance. And for what I saw earlier in the source file, those informations could be updated with the value of the zaptel.conf file. Thanks for your help -- Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Trouble with Realtime
On Fri, 11 Mar 2005 16:38:38 -0600, Nathan Bowyer [EMAIL PROTECTED] wrote: On Fri, 11 Mar 2005 15:04:03 -0600, Matthew Boehm [EMAIL PROTECTED] wrote: You can't have this: [from-sip] switch = Realtime/[EMAIL PROTECTED] The context in your extensions.conf must be different from your Realtime context. Okay, I'll try it, but that doesn't explain why voicemail doesn't work. The extension to access the voicemail is static in extensions.conf. -Matthew Nathan Bowyer wrote: Greetings, I'm having some trouble with the realtime engines. When asterisk loads, everything looks fine, there don't seem to be any problems via notices or anything. Furthermore, cdr_odbc is working, and actively logging my failed call attempts to db through ODBC using the same DSN. unixODBC and the mysql drivers are installed from source. Here are the relevant parts of the config: Extconfig.conf (Under the [settings] section) sipusers = odbc,voip,sip_users sippeers = odbc,voip,sip_users voicemail = odbc,voip,voicemail_users extensions = odbc,voip,extensions_table res_odbc.conf [asterisk] dsn = MySQL-asterisk username = voip password = temp123 pre-connect = yes Under extensions.conf, in the [from-sip] context: switch = Realtime/[EMAIL PROTECTED] Running isql MySQL-asterisk voip pass connects to the DB, and queries return the proper data. I have the following tables in the mysql databases: +--+ Tables_in_voip | +--+ cdr | extensions_table | sip_users| voicemail_users | +--+ In voicemail_users I have an entry for 100101, and in extensions_table I have an extension 520, priority 1 to playback tt-monkeys. Asterisk fails to acknowlege the existence of either. sip_users is blank, and cdr holds the (working) CDR information. In /usr/local/etc/odbc.ini I have: [MySQL-asterisk]] Description = MySQL ODBC Driver Testing Driver = MySQL #Socket = /var/run/mysqld/mysqld.sock Server = 10.10.15.30 User= voip Password= temp123 Database= voip Option = 3 #Port = and odbcinst.ini: [MySQL] Description = MySQL ODBC MyODBC Driver Driver = /usr/lib/libmyodbc3.so FileUsage = 1 UsageCount = 2 If I've missed some relevant part of the configuration, let me know, but I think I got all of it. I'm pretty mistified at the moment, after a few hours of working on it. Oh yes. I also tried a realtime update mailbox 100101 password 1357 from the * CLI, but it errored out. It suggested to check the debug log, but the debug log shows absolutely nothing about Realtime. I've loaded and unloaded app_realtime.so, to no effect. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Looking for an Asterisk Expert/Partner for project
Hello All, How are you all doing today? Good I hope. Well, I have been learning a lot about Asterisk and must say that it is GREAT I have been able to configure and compile various ideas that I have been working on in my test Asterisk PBX and everything worked just fine. The reason for this email to the list is because I have now learned that in order to complete the project that I want to get up and running, will require more time and experience that I can devote to the project at one time. There is much configuring and setup that still needs to be done. I would like to field the question to the list to see if there is anyone or small comapny possibly interested in partnering with me on this project? If so then we can discuss the finer points of the partnership in more detail if you would contact me offline from the list. I know that everyone is extremely busy with their own projects and company goals but I just thought that I would ask anyway to try and get an idea. Have a great day, Lonnie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Droping calls
So, should I just disable busydetect then? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Sábado, 12 de Marzo de 2005 04:13 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Droping calls I have no idea. I live in the USA so I don't normally need busydetect. Anton Krall wrote: Why does busydetect actually drop calls while stile talking? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Viernes, 11 de Marzo de 2005 03:58 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Droping calls Anton Krall wrote: Guys, this is weird.. Today I started having some problems with calls been dropped. Im suing X100p cards (clones) and I have this setting on my zatala fle: [channels] [snip] busydetect=yes busycount=4 Can the echotraining be messing things? Do I need to enable callprogress or something? What do guys think? callprogress and busydetect should both be renamed to randomlydisconnectmycalls because that is what they do. You can REDUCE the number of random disconnects by increasing the value of busycount. There's not anything you can do about disconnects caused by callprogress. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Droping calls
Will give it a try. Thx! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Sábado, 12 de Marzo de 2005 07:15 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Droping calls Guys, this is weird.. Today I started having some problems with calls been dropped. Im suing X100p cards (clones) and I have this setting on my zatala fle: busydetect=yes busycount=4 Try changing busycount to 6 or 8, stop asterisk, and restart. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime does not work yet, ...
Are you running any NAT anywhere? I see that your NAT value is set to '1'. It should be 'yes' or 'never'. That might be your problem. Have you tried adding this user into the sip.conf first to verify that this is truly an ARA problem? -Matthew From: Ronald Wiplinger [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Sat, 12 Mar 2005 07:50:54 +0800 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Realtime does not work yet, ... Matthew Boehm wrote: 2. Run the query inside MySQL cli. How many rows where returned? If none, then its your fault it failed. How do I do that inside of CLI? if you don't know what the MySQL CLI is then you need to stop using mysql. Are you talking about host *CLI or host mysql ? At least I was looking for a way to get the data via *CLI! I see that you are running phpmyadmin, did you run the query that debug spits out? How many rows returned? Did you create the table using the schema on the wiki? Do you have all the columns? Running newest versions of everything? mysql select * from sip_buddies where name='621'; ++--+-+--+---+--+- +-+---+--+--++-+-- -+---+--+--+---+---+-+ +--+--+-+--+-+-++- ---+---++--+---+--+--- --++++ | id | name | accountcode | amaflags | callgroup | callerid | canreinvite | context | defaultip | dtmfmode | fromuser | fromdomain | host| incominglimit | outgoinglimit | insecure | language | mailbox | md5secret | nat | permit | deny | mask | pickupgroup | port | qualify | restrictcid | rtptimeout | rtpholdtimeout | secret| type | username | allow | disallow | musiconhold | regseconds | ipaddr | cancallforward | ++--+-+--+---+--+- +-+---+--+--++-+-- -+---+--+--+---+---+-+ +--+--+-+--+-+-++- ---+---++--+---+--+--- --++++ | 1 | 621 | NULL| NULL | NULL | Demo 621 | yes | inhouse | NULL | rfc2833 | NULL | NULL | dynamic | NULL | NULL | NULL | NULL | [EMAIL PROTECTED] | NULL | 1 | NULL | NULL | NULL | 1 | | 999 | NULL| NULL | NULL | Password | friend | 621 | ulaw;alaw | all | NULL| 0 || yes| ++--+-+--+---+--+- +-+---+--+--++-+-- -+---+--+--+---+---+-+ +--+--+-+--+-+-++- ---+---++--+---+--+--- --++++ 1 row in set (0.00 sec) I believe I use the newest version, since I installed it from the wiki just three days ago! (copy and past) I have only one record in the database (phone 621) bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ASTCC - Regex: How to Country but special City different?
I am trying to figure out a way to add something like: 61 100 pennies (Everything what is not listed below) 61 78150 61 5 130 61 342 180 How could I do these (four) regex? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime does not work yet, ...
sipfriends = mysql,astconf,sip_buddies Yes. Remove that line. This was done a few weeks ago to better split peers/users. sip show users and sip show peers does not show the phone, but Go into sip.conf and enable the 3 RealTime cacheing variables. This will make them show up in the CLI. sip show users and sip show peers still do not show anything. If you don't have RTCacheing on, then this is correct behavior. Try removing both phones from database and put them both into sip.conf and try again. If it works that way then there is a db data problem. As far as the voicemail goes, I'm not sure what happened with the 'addmailbox' script. I've never used it. I just add a new record to the db and the path is created automagically by app_voicemail. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] checking active SIP members of a queue?
hi having a queue with some SIP members, is there a way to check how many of them are connected to asterisk, and if none are, go to a different context? thanks roy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to create channel of type 'IAX2'
On Sat, Mar 12, 2005 at 04:04:29PM +0100, Androtech wrote: Hi all, I'm a newbie and I have a configuration problem with Asterisk. Seems that I'm not able to call an outbound number. I'm quite sure that it is a configuration problem, but I'm not able to find out where is the mistake, even reading several docs to www.voip-info.org. You didn't read all the messages, see: -- Executing Dial(IAX2/[EMAIL PROTECTED]/2, IAX2/3479450772) in new stack Mar 12 16:15:36 WARNING[3149]: chan_iax2.c:2341 create_addr: No such host: 3479450800 Mar 12 16:15:36 NOTICE[3149]: app_dial.c:911 dial_exec_full: Unable to create channel of type 'IAX2' (cause 3) chan_iax2 says you have no section in your iax.conf telling it what 3479450772 is, *therefore* it couldn't create the channel. Always read *all* the messages, especially the first few since they may indicate something that may cause a failure later. FWIW, you probably wanted IAX2/target/3479450772. Hope this helps, -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] checking active SIP members of a queue?
Roy Sigurd Karlsbakk wrote: having a queue with some SIP members, is there a way to check how many of them are connected to asterisk, and if none are, go to a different context? Not at the moment, no. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTCC or should I use something else for different rates, depending on the calling card?
Ronald Wiplinger wrote: Stephen Misel wrote: Ronald Wiplinger wrote: New developments in our business plan make a change necessary. We would like to offer different prices, depending on the user/calling card. How can we use that with ASTCC? or should we use something else? ASTCC allows for multiple brands and different prices for each. I believe the CVS version has my expiration and maintenance fee patches as well. I use the latest ASTCC version, but I do not see how you can use different cards / different prices. Can you explain that for me, please? As I see it: The routes depend on trunks, but the trunks do not care about the cards. bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi Ronald. There should be a section called Brands. -Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] EADS6550 and asterisk - echo on PSTN call
would like to know if some of you have tested asterisk connected to an EADS 6550 analogique PBX (also know as Nexpan50). Our set up is a Dell Optiplex with 1 TDM400 4 FXO, 1 TDM400 4 FXS, (no other card, each of them have their own IRQ) all ports connected to the EADS. We have GS ATA286 as EP. Calling from ATA to ATA give a perfect sound. Calling to PSTN numbers or reverse side, give echo. We can do what we want in zconfig.h (STEVE2, MARK2, MMX, AGGRESSIV_SUPPRESSOR, NOECHOCAN_DISABLE) or zapata.conf (tx gain=-10.0, echocancel=32 ... 256,), test with differents POTS phone, it change nothing. We even didn't notice changes between our various changes in those files (and yes modules where unloaded between each test). Always the same echo. So know we start to doubt that this echo problem is asterisk related but perhaps more to the PBX. That's why we ask if some of you have/had similar setup with this PBX and if there is a solution. You didn't mention what country your in; if you outside the US, be sure to config the TDM-fxo card for your country (eg, line impedance). France. You mention echocancel=32, etc, did you try echotraining=800? Yes. It create a second echo :-( For my TDM-fxo in the US, using the following on each channel works fine: echotraining=800 echocancel=yes echocancelwhenbridged=yes rxgain=5.0 txgain=0.0 Do you have this setup with the standard zconfig.h (MARK2)? I might have missed some of your earlier posts relative to this; just catching up on over 500 emails from this list. I've not had to configure a TDM for non-US support, but I know for an absolute fact (based on 20 years of detailed telephony engineering experience) that you have to config the TDM card for line impedance, etc, for your country. If you've not done that, start there. (Think that's an optional parameter when loading the drivers.) I update asterisk from cvs-head about every two weeks or so, and always stick with default values (including zconfig.h). So, yes I'm using the default echo cancellation, etc. There has not been very many changes associated the the zaptel source code and the TDM-fxo drivers. Certainly not necessary to use the latest cvs-head at all; anything from the last few months should work. My /etc/zaptel.conf is adapted to country: loadzone=fr defaultzone=fr Asterisk stable 1.0.5. If you're telling that I have to pass parameters to module when loading, I checked with modinfo wctdm (at office I have head version) and options I have are those: [EMAIL PROTECTED] asterisk]# /sbin/modinfo -p wctdm debug int loopcurrent int robust int _opermode int opermode string timingonly int lowpower int boostringer int fxshonormode int battdebounce int battthresh int alawoverride int Pardon my ignorance but no one of them remaind me to impedance. And for what I saw earlier in the source file, those informations could be updated with the value of the zaptel.conf file. I believe its the opermode string that needs to be set to a country. Not sure what values are acceptable, but one google result indicated: opermode=Australia as an example. The driver name for the tdm-fxo card/modules has changed to wctdm, so when you look at those google examples, keep that in mind. I pretty sure you need to do the same thing for the TDM-fxs card. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTCC or should I use something else for different rates, depending on the calling card?
Stephen Misel wrote: Ronald Wiplinger wrote: New developments in our business plan make a change necessary. We would like to offer different prices, depending on the user/calling card. How can we use that with ASTCC? or should we use something else? ASTCC allows for multiple brands and different prices for each. I believe the CVS version has my expiration and maintenance fee patches as well. I use the latest ASTCC version, but I do not see how you can use different cards / different prices. Can you explain that for me, please? As I see it: The routes depend on trunks, but the trunks do not care about the cards. Hi Ronald. There should be a section called Brands. Yes, there is a section called Brands! I still do not see it: The routes depend on trunks, but the trunks do not care about the cards. bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to create channel of type 'IAX2'
I do not undertand how to tell to Asterisk to use the X100P to dial an external number instead the internal one. It always calls the internal extension. Could someone give me a valid iax.conf and extension.conf examples files. Regards, chan_iax2 says you have no section in your iax.conf telling it what 3479450772 is, *therefore* it couldn't create the channel. Always read *all* the messages, especially the first few since they may indicate something that may cause a failure later. FWIW, you probably wanted IAX2/target/3479450772. Hope this helps, -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk and Broadvoice Outgoing Again :(
Hi,
I tried that but same error
Specially I didn't find people posting about Bad Request or Unknown Dialog
-- Got SIP response 400 Bad request back from 147.135.8.128
-- SIP/5092321848-ccd4 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Congestion(SIP/502-f815, 5) in new stack
-- Got SIP response 400 Bad request back from 147.135.8.128
-- Got SIP response 481 Unknown Dialog back from 147.135.8.128
On Friday 11 March 2005 02:30, Dan Weber wrote:
Here is my sip.conf:
===
register =
[EMAIL PROTECTED]::[EMAIL PROTECTED]/broadvoi
ce
[]
type=peer
user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=
secret=
username=
insecure=very
context=default
authname=
dtmfmode=inband
dtmf=inband
canreinvite=no
Contact information must not change between register and call. Whats
happening here is that when you register its [EMAIL PROTECTED], however,
when you call, its [EMAIL PROTECTED] Change the extension of the
register to match your phone number.
Dan
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Spam detection software, running on the system zeus.avanzada7.com, has
identified this incoming email as possible spam. The original message
has been attached to this so you can view it (if it isn't spam) or label
similar future email. If you have any questions, see
the administrator of that system for details.
Content preview: Here is my sip.conf:
=== register =
[EMAIL PROTECTED]::[EMAIL PROTECTED]/broadvoice
[] type=peer user=phone host=sip.broadvoice.com
fromdomain=sip.broadvoice.com fromuser= secret=
username= insecure=very context=default
authname= dtmfmode=inband dtmf=inband canreinvite=no
[...]
Content analysis details: (0.1 points, 5.0 required)
pts rule name description
--
-- 0.1 FORGED_RCVD_HELO
Received: contains a forged HELO
--
With regards,
Vicky Shrestha
System Director
WorldLink Communications
Jawalakhel , Kathmandu, Nepal
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Re: [Asterisk-Users] asterisk and Broadvoice Outgoing Again :(
It doesn't try to authenticate the incoming call. On Friday 11 March 2005 03:56, Randy Johnson wrote: What does insecure=very do? Dan Weber wrote: Here is my sip.conf: === register = [EMAIL PROTECTED]::[EMAIL PROTECTED]/broadvo ice [] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser= secret= username= insecure=very context=default authname= dtmfmode=inband dtmf=inband canreinvite=no Contact information must not change between register and call. Whats happening here is that when you register its [EMAIL PROTECTED], however, when you call, its [EMAIL PROTECTED] Change the extension of the register to match your phone number. Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- With regards, Vicky Shrestha System Director WorldLink Communications Jawalakhel , Kathmandu, Nepal ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk and Broadvoice Outgoing Again :(
Thanks, I have that already in my /etc/hosts But it's still not working :( On Saturday 12 March 2005 03:48, Rich Adamson wrote: For everyone that's trying to get BV to work, you'all might want to edit your /etc/hosts file and insert something like: 147.135.8.128 sip.broadvoice.com This was a requirement from way back and I've since discontinuted BV for a different provider, but seems as though of all the suggestions posted in recent weeks, few mention the above. After editing /etc/hosts, there is no need to reboot, etc. The contents are read dynamically. Then make sure that your contexts and extensions.conf use sip.broadvoice.com in them. They did have four different servers at one time (with four different IP's), but if you stick with one (like the above) and play with the other parameters to get it to work, then you can change servers at a later time. As one more comment, any changes that you make to sip.conf or extensions.conf associated with trying to make BV work, don't forget to stop and restart asterisk. Don't rely on a reload as it does not reread all parameter changes. I can't make outgoing calls via Broadvoice. I have tried each and every configuration that was posted to list previously. I am able to receive incoming calls fine. I get the following in asterisk console: = asterisk*CLI show version Asterisk CVS-HEAD-03/10/05-22:51:28 built by [EMAIL PROTECTED] on a i686 running Linux asterisk*CLI -- Executing Dial(SIP/502-c147, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- Got SIP response 400 Bad request back from 147.135.8.128 -- SIP/-19dd is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing Congestion(SIP/502-c147, 5) in new stack == Spawn extension (vicky, 0018086749157, 2) exited non-zero on 'SIP/502-c147' -- Got SIP response 400 Bad request back from 147.135.8.128 -- Executing Dial(SIP/502-8efd, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- Got SIP response 400 Bad request back from 147.135.8.128 -- SIP/-4bf5 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing Congestion(SIP/502-8efd, 5) in new stack == Spawn extension (vicky, 008086749157, 2) exited non-zero on 'SIP/502-8efd' -- Got SIP response 400 Bad request back from 147.135.8.128 -- Got SIP response 481 Unknown Dialog back from 147.135.8.128 Here is my sip.conf: === register = [EMAIL PROTECTED]::[EMAIL PROTECTED]/broadvoi ce [] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser= secret= username= insecure=very context=default authname= dtmfmode=inband dtmf=inband canreinvite=no -- With regards, Vicky Shrestha System Director WorldLink Communications Jawalakhel , Kathmandu, Nepal ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- -- With regards, Vicky Shrestha System Director WorldLink Communications Jawalakhel , Kathmandu, Nepal ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Advanced conference features, meetme2?
Hi, I have been playing about with meetme as a conference bridge, and find it lacking in some features which I believe are out their somewhere. Viewing this wiki page http://www.voip-info.org/wiki-Asterisk+MeetMe2+Design it looks like a plan happened, but where is meetme2 at now? Things like recording a conference, allowing callers to adjust volume, allowing the conference to be locked, having the users name recorded before entering, and then played back to other callers on entrance etc etc. Are these things available now, or would they require development. Regards C attachment: winmail.dat___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How do I pick up a trailing number in extensions.conf?
Hi,
my SIP provider sends me all the numbers that are dialed, i.e. when a
number is appended to the phone number proper, it gets appended to the
incoming number, like (my number is 0123456789, and I append a 5):
To: sip:[EMAIL PROTECTED];tag=as1174b008
Question is, how can I use the trailing number in extensions.conf? This is
ideal for a direct dial through to an extension in my ISDN PBX (which I use
as an 8-way ISDN-analog adapter ;-)
I tried something like
exten = sip,1,Goto(isdn,1${EXTEN:-1},1)
where sip is the extension name in
register = NNN:[EMAIL PROTECTED]/sip
But apparently ${EXTEN:-1} is p because... how do I do that?
TIA!
--
On two occasions I have been asked [by members of Parliament!], `Pray,
Mr. Babbage, if you put into the machine wrong figures, will the right
answers come out?' I am not able rightly to apprehend the kind of
confusion of ideas that could provoke such a question.
-- Charles Babbage
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[Asterisk-Users] ANNOUNCEMENT: Updates for app_cbmysql and MeetMe2 gui (out of tree modules)
New PHP web interface at www.fitawi.com/Asterisk There is also a sample cbmysql config file and the database Tables description. -New in the interface: Added schedule conflict detection for Add and Update conference functions. -New in app_cbmysql: No changes -ToDo in the interface: Mouse-over support in Monitor conference for Caller-Id -ToDo in app_cbmysql: Nothing planned -ToDo in app_meetme2 Use Caller-ID for user_id An interface to mute or boot callers kind of needs this Port recent MeetMe bugfixes and enhancements Add conference termination code Add conference ending warning Original source: App_MeetMe2 http://www.areski.net/asterisk-meetme/about.php?s=0 App_CBMysql http://www.mithotech.com/asterisk/ Thanks and enjoy, Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Parked Call
Thanks for the reply. Here is my sip.conf [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default disallow=all allow=ulaw dtmfmode=rfc2833 register = [EMAIL PROTECTED]:password:[EMAIL PROTECTED]/ canreinvite = no [sip.broadvoice.com] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=XX2061 secret=PASSWORD username=XX2061 insecure=very context=mainmenu authname=XX2061 dtmfmode=rfc2833 dtmf=rfc2833 canreinvite = no [] type = friend host = 192.168.1.111 context = out allow = ulaw canreinvite=no extension = regexten = It still does not work. In extensions.conf I have include = parkedcalls and still no go. Thanks again for the help, justin Guido Hecken wrote: Try using dtmfmode=rfc2833 in your sip.conf. It should work... Hope, this could help. Guido Hecken I have a question, I am unclear on how to park a call. I know that you are supposed to be able to press # and then transfer the call to extension 700. However, * doesn't seem to be graping the dtmf. I am using dtmfmode=inband. Asterisk is in the media path as well. Thanks in advance Justin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Parked Call
Hello Justin, dtmfmode should be inband for broadvoice either way because that's what they support. Now for the extensions.conf do you have: exten = ,1,Dial([SIP|IAX2|..]/something, timeout, t) -- t for transfers? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Justin Ramsey Sent: Saturday, March 12, 2005 2:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Parked Call Thanks for the reply. Here is my sip.conf [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default disallow=all allow=ulaw dtmfmode=rfc2833 register = [EMAIL PROTECTED]:password:[EMAIL PROTECTED]/ canreinvite = no [sip.broadvoice.com] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=XX2061 secret=PASSWORD username=XX2061 insecure=very context=mainmenu authname=XX2061 dtmfmode=rfc2833 dtmf=rfc2833 canreinvite = no [] type = friend host = 192.168.1.111 context = out allow = ulaw canreinvite=no extension = regexten = It still does not work. In extensions.conf I have include = parkedcalls and still no go. Thanks again for the help, justin Guido Hecken wrote: Try using dtmfmode=rfc2833 in your sip.conf. It should work... Hope, this could help. Guido Hecken I have a question, I am unclear on how to park a call. I know that you are supposed to be able to press # and then transfer the call to extension 700. However, * doesn't seem to be graping the dtmf. I am using dtmfmode=inband. Asterisk is in the media path as well. Thanks in advance Justin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Vonage a provider?
Don't sweat it - it just so happens that you came into the fray just moments after this list has had a big long drawn out argument about newbie etiquete (sp?). You've just managed to get caught in the middle. Don't let it be indicative of how everyone feels, and don't let it scare you away... :) regards, Paul - Original Message - From: Frank Abernathy [EMAIL PROTECTED] To: 'C F' [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, March 11, 2005 3:49 PM Subject: RE: [Asterisk-Users] Vonage a provider? Sorry to have caused such a ruckus. It was not my intent to 'anger' someone with a noob question. I did a look-up on Google, hence me getting the information about this mail list. I am sorry that I am not a Google guru like you, so that my look-up did not get me the information I needed, so that I had to 'bother' an actual person and not some search engine. I guess some people were never noobs. :) Regardless, thank you for your response and information. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Friday, March 11, 2005 3:16 PM To: Wiley Siler; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Vonage a provider? On Fri, 11 Mar 2005 13:56:20 -0700, Wiley Siler [EMAIL PROTECTED] wrote: I don't feel I am mistreating you in asking you not to dump on a noob. Even if you do not think he is a noob and he is just lazy. You wrote: Why answer? because I don't want this to happen again. But I dont' care to help him/her or even you. Sorry this should have been: Why answer? because I don't want this to happen again. But I don't mind to help him/her or even you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.7.1 - Release Date: 3/9/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.7.1 - Release Date: 3/9/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] EADS6550 and asterisk - echo on PSTN call
administrator tootai wrote: If you're telling that I have to pass parameters to module when loading, I checked with modinfo wctdm (at office I have head version) and options I have are those: [EMAIL PROTECTED] asterisk]# /sbin/modinfo -p wctdm debug int loopcurrent int robust int _opermode int opermode string timingonly int lowpower int boostringer int fxshonormode int battdebounce int battthresh int alawoverride int Pardon my ignorance but no one of them remaind me to impedance. And for what I saw earlier in the source file, those informations could be updated with the value of the zaptel.conf file. You need to add opermode=FRANCE If you are using Didium FXS modules as well, with non 600 ohm phones, i.e. European TBR21 standard, you should add fxshonormode=1 as well. This will set the FXS impedance to whatever is specified in the opermode = parameter. You can confirm this afterwards by checking dmesg for an entry showing: Module 0: Installed -- AUTO FXO (FRANCE mode) Regards, Richard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] digium card
On March 11, 2005 10:49 pm, M.N.A.Smadi wrote: does any body know what are the physical dimension of a digium care 400pm for example? It's a half-length PCI card. It's maybe an inch or so longer than the PCI slot itself. It is full-height though (i.e. the circuit board is as tall as the back slot). As far as width -- you can fit it right beside any other PCI card so long as the card it's beside does not have a heatsink or fan. It's tight that way. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Where to download the asterisk-oh323?
Dear ALL: Where can I find the oh323 module on CVS or anywhere? I want to implement the SIP(ser) to Asterisk to H323(gnugk). Thank you. Best Regards Charles ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Incoming echo cancel
On March 11, 2005 04:51 pm, Eric Wieling wrote: echocancel=yes echotraining=yes or 600 or 800 I absolutely *despise* echotraining. A half second (or in your case 8/10 of a second) delay before hearing anything is unacceptable in almost all situations. Maybe if you've got a physical disability which prevents you from picking up the phone and saying hello in under a second (arthritic joints, middied cognition?)... it's unacceptable. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoipJet Terms of Service
On March 11, 2005 10:49 am, John Goerzen wrote: * Then there's the NDA: People are specifically prohibited from telling anyone that they use VoipJet, including end users. Also, we can't tell people what we pay for it, even though the prices are right there on their website. * Then there is this one: The Customer agrees not to undertake any action . . . that would harm VoipJet . . . in any way, including financially. So, if I got crappy service from VoipJet and blogged about it, and thus cost them business, even if the NDA didn't get to me, this would, even if my account was completely accurate. How sickening. You can put any silly thing you want in an agreement. A judge would puncture any NDA clause just as they'd puncture an illegal clause in a EULA or any other contract. Oracle has these stupid types of clauses in their contracts for evaluations too. The only reason people honor them is because if they don't they'll never have another opportunity to evaluate the software and the bad karma would follow them around for years. (Think of online mags and the like.) -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTCC and NuFone billing is different!!
On March 11, 2005 11:13 am, Ronald Wiplinger wrote: I have ASTCC installed, and compare it with NuFone, however, I find that the billing of NuFone is always a few secondes more (6 to 24 seconds) Does anybody has an explanation / solution for it? Have you emailled [EMAIL PROTECTED] and asked them? IIRC their billing rounds to the nearest 1/10 min (i.e. nearest 6 seconds) -- maybe it's 15, I can never remember. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTCC and NuFone billing is different!!
NuFone service bills in industry standard billing increments, which are: six (6) seconds for the US48, sixty (60) seconds to Mexico and fifteen (15) seconds to the remainder of the world. From: http://www.nufone.net/tac.html ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No ringback over IAX - LiveVoip
Title: RE: [Asterisk-Users] No ringback over IAX - LiveVoip Excellent thing to hear. I am glad there are positives on this site as well as teh warnings. Now to get the ringback issue resolved Using m switch to get MOH is OK but there has to be alogical reason this is occuring adn a way to resolve. Thanks, Wiley From: [EMAIL PROTECTED] on behalf of Rich AdamsonSent: Sat 3/12/2005 5:49 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] No ringback over IAX - LiveVoip Grett. This should be loads of fun then... 8( I have noticed what I can only describe a negative undertone with several VoIP poviders. Not an easy customer? We don't want you. Things like that. The LiveVoIP website is in fact like that. There are several places on the site that just flat out say "are you customer type x?, we don't want you then". Not my way of doing business but to each their own. I guess as long as they service my account and provide a good voip connection it won't mean much to me.There are more then a few folks using * that try various itsp serviceswithout a clue as to how to make things work, and/or with unrealisticexpectations. I happen to like their "no-nonsense" approach on theirweb site of getting your attention. It sort of resembles some of thepostings on this list relative to 'did you try to look for doc'.Overall, I'd give livevoip high marks in both quality and service. Let'shope they can keep it up. :)___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Best DB
On March 10, 2005 07:14 pm, Giudice, Salvatore wrote: I vote for MySQL. PostgreSQL is fine, but MySQL handles much better under extreme load. MySQL is also usually touted as being generally You *gotta* be kidding me. MySQL can't hold a candle to PostgreSQL for high load, high volume or complex queries. MySQL is great for simple selects and light duty use but you will have to introduce clustering and failover much sooner for MySQL than you ever will for Postgres. As far as speed goes, MySQL's speed falls down *very* quickly once you start using anything more than simple SELECTs. Throw in some joins, some ordering and complex clauses and it grinds to a crawl. functionality in stored procedures, cursors, and views. In terms of support, you can get support from MySQL directly, while PostgreSQL means you have to turn to mailing lists. It's really your preference depending There are plenty of companies to help you with PostgreSQL, http://www.commandprompt.com being the most obvious choice (they will sell you PostgreSQL with a support license.) supporting open source in house. Lastly, be aware that MySQL is distributed under the GNU license with a commercial rider for derivative works and PostgreSQl is a BSD license. MySQL's constant licensing issues are the biggest reason why it's not natively supported in Asterisk! Please, please, please get your facts straight. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zapping around
Dear list, I am trying to learn how to use Zap-things in Asterisk. While loading Asterisk verbosely I get this error: [chan_zap.so]Warning, flexibel rate not heavily tested! = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found Mar 12 17:19:01 WARNING[5563]: chan_zap.c:763 zt_open: Unable to open '/ dev/zap/channel': No such file or directory Mar 12 17:19:01 ERROR[5563]: chan_zap.c:6208 mkintf: Unable to open channel 1: No such file or directory here = 0, tmp-channel = 1, channel = 1 Mar 12 17:19:01 ERROR[5563]: chan_zap.c:9155 setup_zap: Unable to register channel '1-2' Mar 12 17:19:01 WARNING[5563]: loader.c:345 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' Mar 12 17:19:01 WARNING[5563]: loader.c:440 load_modules: Loading module chan_zap.so failed! [EMAIL PROTECTED]:~ Ouch ... error while writing audio data: : Broken pipe Now, modprobing zaptel is ok: [EMAIL PROTECTED]:~ sudo /sbin/modprobe zaptel [EMAIL PROTECTED]:~ Same with wcfxo, wcfxs [EMAIL PROTECTED]:~ sudo /sbin/modprobe wcfxo [EMAIL PROTECTED]:~ sudo /sbin/modprobe wcfxs [EMAIL PROTECTED]:~ although here I do have a first question. The card I am using is a TDM400 with two FXS module, an empty slot and an FXO module. The wiki mentions a wctdm module that I do not find (modprobing it just fails). Am I missing something, or can I use the older set of kernel modules? [info: I did get the 1.0.6 zaptel, libpri and asterisk archives from the Digium site; I did compile everything under SUSE 9.2, thus with a stock 2.6.8-24-default kernel; I did use the make linux26 command in the install process of zaptel. ] Now the card seems to react to my fiddling: the three green leds corresponding to the installed module positions do turn on as soon as I type the wcfxo or wcfxs modprobe command. The zaptel config file is as follows: ** zaptel.conf ** the stock file as generated by the compile process, with the addition of these lines # edited by aaberga % 12.3.05 loadzone = us defaultzone = us fxsks = 1-2 #fxoks = 4 ** zaptel.conf ** The zapata.conf file is as follows: ** zapata.conf ** the stock file as generated by the compile process, with the addition of these lines ; edited by aaberga % 12.3.05 ;signalling = fxs_ks ;context = incoming ;channel = 4 signalling = fxo_ks context=internal channel = 1-2 ** zapata.conf ** I am obviously missing and/or misdoingsomething; can anybody help? Thanks in advance Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Tracking/Billing Incoming Outgoing Minutes?
Hello, I found a way to do a callback service using call files however, only call leg B gets recorded in the CDR so I would only be able to accurately bill for one leg? Does anyone have any suggestions on how to get numbers for both call legs? Or a way to bridge two seperate SIP calls and have both tracked? Jess ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] InterVivo and MusicOnHold()
Hi All, I've been trying for a while to get * to play MusicOnHold with my SIP connection. I can hear it when I call a test extension from my local X-Lite phone, but when I dial in via InterVivo, I just hear silence. I have a Gentoo box with kernel 2.4.28-gentoo. I have no sound card or speakers on the box, it's in a cupboard. I have uncommented the lines in musiconhold.conf. I am trying to use the following extensions [inbound-calls] exten = s,1,Dial(SIP/07X,20,m) [voip] exten = test,1,Goto(inbound-calls,s,1) Dialing test from X-Lite works correctly, and dialling in diverts to the mobile, but with silence. PlayBack() with GSM files works okay. Is there something special I need to do with InterVivo to get it to work? Thanks Jamie SIP.CONF attached: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind SIP channel to context = inbound-calls ; Default context for incoming calls srvlookup = yes ; Enable DNS SRV lookups on outbound calls disallow=all allow=gsm allow=ulaw allow=alaw allow=ilbc externip = register = 0207043:[EMAIL PROTECTED] realm = voip.project76.net localnet = 41.0.0.0 localmask = 41.240.0.0 nat = yes ;outbound calls go here [sip-with-london-number] type=friend secret= username=0207043 host=sip.intervivo.net insecure=very fromuser=0207043 fromdomain=sip.intervivo.net ;soft phone client [jamie] type=friend secret= host=dynamic nat=yes username=jamie disallow=all allow=gsm allow=ulaw allow=alaw context=voip -- Visit our Little Britain microsite: http://little.britain.project76.tv/welcome.php You can now contact us at local call rates(*) via our NEW number: 0845 226 9157. (*) May not be included in your provider's call allowance. Check with provider for call costs. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Advanced conference features, meetme2?
On Sat, 12 Mar 2005, C. Tomlinson wrote: I have been playing about with meetme as a conference bridge, and find it lacking in some features which I believe are out their somewhere. Viewing this wiki page http://www.voip-info.org/wiki-Asterisk+MeetMe2+Design it looks like a plan happened, but where is meetme2 at now? Things like recording a conference, allowing callers to adjust volume, allowing the conference to be locked, having the users name recorded before entering, and then played back to other callers on entrance etc etc. Are these things available now, or would they require development. At least some of these are already in meetme in cvs head. Some can be implemented in the dialplan. Some I guess are not available yet. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Signaling on PRI channels
On Sat, 12 Mar 2005, Laurent Tostain wrote: Hi, We did an interconnection with our carrier few days ago. But, I noticed that there was a signaling problem on our trnuk. In fact, Asterisk indicates that the call is answered when we received ALTERTING message from our carrier. This is PRI debug logs : [...] This is the zapata.conf file : [channels] [...] callprogress=yes I guess this could be the culprit. You do not want audio call progress analysis on a pri. Perhaps that is what is confusing Asterisk. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail to UK mobile
Hi Can u let me know how you are doing this, please Iqbal On 3/12/2005, Rafal Kaniewski [EMAIL PROTECTED] wrote: What issues/options are there when forwarding voicemail to uk mobile voicemail? ta Rafal Kaniewski -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.7.2 - Release Date: 11/03/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Broadvoice outgoing problems
Hello All,
I'm just getting into *, and trying to use a Broadvoice account. It
works inbound, but Outbound fails no matter what sip.conf parameters I try.
From the recent posts here I think it could be:
A bad CVS release - I will try to download and build from a new one
Broadvoice not challenging and/or Asterisk not responding with an
Authorization: in the INVITE header.
I am able to call outbound with the account using an X-Lite Soft phone on my
Windows PC, so I know my username and password settings are correct. I did an
Ethereal trace on the softphone call setup and compared it to a trace of the
call attempt from *. I noticed that the softphone sends an Authorization: line
in the INVITE header, while * does not.
I am registered successfully:
*CLI sip show registry
HostUsername Refresh State
sip.broadvoice.com:5060 UU 11 Registered
Here is what I am getting in SIP debug:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.9:5060;branch=z9hG4bK1682a548
From: Jay D. Carter sip:[EMAIL PROTECTED];tag=as741ea376
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 12 Mar 2005 21:56:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 156
v=0
o=root 4202 4202 IN IP4 192.168.0.9
s=session
c=IN IP4 192.168.0.9
t=0 0
m=audio 16022 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
(no NAT) to 147.135.12.128:5060
-- Called [EMAIL PROTECTED]
Sip read:
SIP/2.0 100 Trying
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
From: Jay D. Carter sip:[EMAIL PROTECTED];tag=as741ea376
To: sip:[EMAIL PROTECTED]
Via: SIP/2.0/UDP 192.168.0.9:5060;branch=z9hG4bK1682a548;received={my public
IP};rport=18992
Content-Length:0
7 headers, 0 lines
Sip read:
SIP/2.0 403 Forbidden
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
From: Jay D. Carter sip:[EMAIL PROTECTED];tag=as741ea376
To: sip:[EMAIL PROTECTED];tag=adeg
Via: SIP/2.0/UDP 192.168.0.9:5060;branch=z9hG4bK1682a548;received={my public
IP};rport=18992
Content-Length:0
7 headers, 0 lines
Transmitting:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.9:5060;branch=z9hG4bK1682a548
From: Jay D. Carter sip:[EMAIL PROTECTED];tag=as741ea376
To: sip:[EMAIL PROTECTED];tag=adeg
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 147.135.12.128:5060
Mar 12 16:56:32 WARNING[4202]: chan_sip.c:6829 handle_response: Forbidden -
wrong password on authentication for INVITE to 'Jay D. Carter sip:[EMAIL
PROTECTED];tag=as741ea376'
-
My sip.conf:
[bv]
type=peer
host=sip.broadvoice.com
outboundproxy=proxy.chi.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=UU
username=UU
secret=mysecret
;authname=UU
;authuser=UU
context=incoming
port=5060
;Disable canreinvite if you are behind a NAT
canreinvite=no
insecure=very
dtmfmode=inband
disallow=all
allow=ulaw
Note: The commented lines in the sip.conf have been tried - same result
Hosts file has:
147.135.12.128 sip.broadvoice.com
..I have also tried without any hosts entry.
I'd really appreciate the sanity check any assistance you all can provide.
Thanks!
JDC
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RE: [Asterisk-Users] what is best free softphone.
Pulver.communicator (FWD) ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of FCG ZHAO Zigang Sent: Freitag, 11. Mrz 2005 06:17 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] what is best free softphone. I use xlite , but it isn't support video when it is free. who used better softphone ? Thank u. Best Regards Zhao Zigang Alcatel Shanghai Bell Co., LTD *:388,NingQiao Rd.,Shanghai 201206 *:086-21-50554550-7762 *:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Best DB
On Fri, Mar 11, 2005 at 01:56:47PM -0500, Giudice, Salvatore wrote: As for the production recommendation you refer to, I would respectufully disagree. If you are an enterprise comapny looking to deploy an open-source DB, you will pick the one that has an established support company to contract with. So, 'NO': postgreSQL is not recommended for production environments. MYSQL AB provides enterprise class support. PostgreSQL support consists of contracting with mom and pop support shops, mailing lists, and irc. That simply will not be acceptable for the enterprise user. You know, you probably shouldn't rely on Linux (the kernel) as there is no company behind it. I also wonder what is your source for support for Apache. What about PHP? is Zend your sole source of support there? BTW: the fact that the MySQL stuff is in add-ons is also because Asterisk is about as strict as MySQL regarding the license. But you may also be interested in reading http://lists.alioth.debian.org/pipermail/pkg-voip-maintainers/2005-February/001301.html -- Tzafrir Cohen | New signature for new address and | VIM is http://tzafrir.org.il | new homepage | a Mutt's [EMAIL PROTECTED] || best ICQ# 16849755 | Space reserved for other protocols | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: DISREGARD!![Asterisk-Users] Broadvoice outgoing problems
... I just tried again after removing my hosts file entry (again) and outbound is now working! I had taken it out before, but I think I was getting a different error at the time. Sometimes it seems like asking for help is itself a cure! Thanks anyway! JDC ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CNAM for Asterisk
I am working for Accudata Technologies. We provide CNAM via http request or raw TCP/IP connection. We would like to provide the same capability to Asterisk. I installed Asterisk on Fedora 2.0 and did reading about AGI and AGI application at http://www.voip-info.org/tiki-index.php?page=Asterisk+AGI, http://www.voip-info.org/wiki-Asterisk+Manager+API, and The Asterisk Handbook Version2 but still not clear on how to interface a new application to Asterisk. At first I would like to interface a simple client application in C that connects to an IP and Port number. The application receives a phone number from Asterisk (???), connects to our server, sends the phone number, receives a caller name from the server, sends the name to Asterisk (???). Thank you for your help. Kevin N. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Video Conference
Hi All; I know Asterisk can support video calls over sip or h323 but I need to know if it can be used in Video Conferencing? Can I use "meet me" for that purpose? Regards Mohammad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Best DB
On Fri, Mar 11, 2005 at 04:25:59PM -0500, David Filion wrote: Maybe I miss read, but doesn't the licensing of the newer releases of MySQL require companies to purchase a license? No. The license is GPL. Originally it was LGPL for the client libraries but this got changed recently. So you have a number of options: 1. Use the GPL libraries, and use the code internally only. As long as you don't distribute your code the GPL imposes no restrictions on your code. 2. Use the GPL libraries and allow your internal code to be exposed 3. Pay MySQL AB for a license. This would mean that while it is open source, it is not free as in beer. This does not mean it is not a good DB, just that there may be more that just the costs of a support contract involved. This is why most distros still ship the last version before the license change. As for support, check out http://techdocs.postgresql.org/companies.php. Debian keeps both 4.0 and 4.1 . Fedora's rawhide now has 4.1, Latest Mandrake has 4.1. -- Tzafrir Cohen | New signature for new address and | VIM is http://tzafrir.org.il | new homepage | a Mutt's [EMAIL PROTECTED] || best ICQ# 16849755 | Space reserved for other protocols | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CNAM for Asterisk
Kevin Nguyen wrote: Thank you for your help. Kevin N. I already replied to your first message with a great deal of information; did you not receive it? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] after *-1.0.6 upgrade error: vm_execmain: Unable to read password
After upgrading to asterisk 1.0.6 on Gentoo when I try to log-in to check the voice mail I get: app_voicemail.c:3389 vm_execmain: Unable to read password -- #Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Asterisk-Users Digest, Vol 8, Issue 88
These allow and disallow work with NuFone for me
disallow=all
allow=ulaw
allow=alaw
allow=gsm
Jeff
Message: 11
Date: Fri, 11 Mar 2005 11:15:51 +0100
From: Edward Banfa [EMAIL PROTECTED]
Subject: [Asterisk-Users] NuFone Configuration [problem]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=us-ascii
Hello,
I am trying to configure the my asterisk box here with the following
**iax.conf***
[NuFone]
type=peer
host=switch-1.nufone.net
secret=xx
***extensions.conf:***
exten = _1NXXNXX,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
exten = _011N.,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
I have a couple of Xlite softphones and 2 analogue phones connected to a
mediatrix 1102 connected to our lan. The mediatrix talks sip to the
asterisk
box on the lan. We are running asterisk on FC3 .
SOFTPHONES[XLITE] ---SIP-- ASTERISKIAX---NUFONE[ASTERISK]
ANALOGPHONES---MEDIATRIX_1102---SIP---ASTERISK---IAX---NUFONE[ASTERISK
]
Well the problem goes something like this.
1) I can dial a number form the softphones and when the call is answered
I
can hear the user on the other end but the user can't hear me
2) I can dial a number from the analog phones (via mediatrix tru to
asterisk)(the mediatrix is properly registered with our asterisk box)
and
when the call is answered both ends can't hear a word, its just silent.
I think I am having a codec problem here. What am I doing wrong. We
would
sincerely appreciate any help/pointers.
Thank you all
Edward Banfa
**EXTENSION.CONF***
[general]
static=yes
[from-sip]
exten = 100,1,Dial(SIP/edward,20)
exten = 100,2,Hangup
exten = 101,1,Dial(SIP/phone1,20)
exten = 101,2,Hangup
exten = 102,1,Dial(SIP/phone2,20)
exten = 102,2,Hangup
exten = _1NXXNXX,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
exten = _011N.,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
*IAX.CONF*
[general]
port=5036
bind=0.0.0.0
bandwidth=low
disallow=lpc10
[NuFone]
type=peer
host=switch-1.nufone.net
secret=xx
disallow=all
allow=ilbc
allow=gsm
allow=ulaw
disallow=all
allow=ulaw
allow=alaw
allow=gsm
**SIP.CONF*
[general]
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
[edward] ;My Xlite softphone
type=friend
host=dynamic
secret=pass-da-word
context=from-sip
callerid=edward 100
mailbox=100
disallow=all
allow=gsm
allow=ulaw
allow=alaw
allow=ilbc
allow=g726
[phone1] ;First analog phone connected to mediatrix
type=friend
host=dynamic
secret=pass-da-word
context=from-sip
callerid=phone1 101
mailbox=101
disallow=all
allow=gsm
allow=ulaw
allow=alaw
allow=ilbc
allow=g726
[phone2] ;Second analog phone connected to mediatrix
type=friend
host=dynamic
secret=pass-da-word
context=from-sip
callerid=phone2 102
mailbox=102
disallow=all
allow=gsm
allow=ulaw
allow=alaw
allow=ilbc
allow=g726
--
Message: 12
Date: Fri, 11 Mar 2005 15:57:38 +0530
From: Jagan Mohan [EMAIL PROTECTED]
Subject: [Asterisk-Users] Load Balancing b/w 2 asterisk servers using
SIP load balancer
To: Asterisk asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=US-ASCII
Hi,
I'm trying to do load balancing between 2 asterisk servers using SIP
load balancer, provided by http://www.vovida.org
I used the following options on lbproxy, but I get the below message
continuously.
./lbProxy -name seneca -reqPort 5060 -respPort 5061 -proxy A1 -proxy A2
No proxies are up - can not send message to anyone
Xlite is not able to register to the asterisk server.
Is there anything which needs to be tweaked on Asterisk side to get this
working? Please help.
Thanks,
Jagan
--
Message: 13
Date: Fri, 11 Mar 2005 11:31:29 +0100
From: Vledder, Hans [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Asterisk and USB ISDN controllers ...
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
m
Content-Type: text/plain; charset=iso-8859-1
Hi Steve,
Since you don't mention what USB ISDN adapter specifically you are
thinking about, what do you think we will be able to tell you.
All I know about the adapter is what I've told you. It's a USB
Colognechip
based ISDN controller - probably HCF-USB based. It's supported by Linux,
but
there's no info on access to B and D channels.
Regards,
Hans
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steven
Critchfield
Sent: Thursday, March 10, 2005 6:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk and USB ISDN controllers ...
On Thu, 2005-03-10 at 18:13 +0100, Vledder, Hans wrote:
Guys,
I am planning on building a small SIP PBX with a single ISDN line.
Currently
I am looking into the specs of a very tiny barebone system that has an
option Colognechip base ISDN
[Asterisk-Users] Sjphone call quality: free phone vs. commercial
Hello, Could anyone say if there is any significant boost in voice quality with the commercial SJPhone (payed for) vs. their free version? Also, any reports on SIPPS free vs commercial? It is worth to buy the licenced SJPhone/SIPPS to increase the voice quality (they want $95 for it, pretty expensive for a softphone)? Thanks, Roman Zhovtulya -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Sonntag, 13. März 2005 00:05 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] CNAM for Asterisk Kevin Nguyen wrote: Thank you for your help. Kevin N. I already replied to your first message with a great deal of information; did you not receive it? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk with Skype
Hi, Does anyone know if it's possible to hook an asterisk PBX up to skype? And if so, any config examples? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] checking active SIP members of a queue?
You mean that if on a certain queue, your agents are using SIP or IAX phones, and you want to do a check so that when a cllers tryies to get into the queue, if no agent is logged in, do something else with the caller instead of hanging up? Is that what you are trying to do? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Sábado, 12 de Marzo de 2005 10:44 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] checking active SIP members of a queue? Roy Sigurd Karlsbakk wrote: having a queue with some SIP members, is there a way to check how many of them are connected to asterisk, and if none are, go to a different context? Not at the moment, no. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Advanced conference features, meetme2?
Conference lock and member name been recorded and announced when they get in and out of a conference is already available. Check the wiki and look for meetme, you will see they have some parametes like m,a,s that will help you control this features. Anton Krall _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C. Tomlinson Sent: Sábado, 12 de Marzo de 2005 12:31 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject:[Asterisk-Users] Advanced conference features, meetme2? Hi, I have been playing about with meetme as a conference bridge, and find it lacking in some features which I believe are out their somewhere. Viewing this wiki page http://www.voip-info.org/wiki-Asterisk+MeetMe2+Design it looks like a plan happened, but where is meetme2 at now? Things like recording a conference, allowing callers to adjust volume, allowing the conference to be locked, having the users name recorded before entering, and then played back to other callers on entrance etc etc. Are these things available now, or would they require development. Regards C File: ATT00137.txt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk with Skype
Never mind.. answered my own question looks like their is a bounty on the ability to do this :P On Sat, 12 Mar 2005 18:28:00 -0500, Matt [EMAIL PROTECTED] wrote: Hi, Does anyone know if it's possible to hook an asterisk PBX up to skype? And if so, any config examples? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Advanced conference features, meetme2?
Title: RE: [Asterisk-Users] Advanced conference features, meetme2? Id like to know this to, Im prepared to kick in $50 to start off a bounty if no one else has done so. Cheers, Dean _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of C. Tomlinson Sent: Saturday, March 12, 2005 1:31 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Advanced conference features, meetme2? Hi, I have been playing about with meetme as a conference bridge, and find it lacking in some features which I believe are out their somewhere. Viewing this wiki page http://www.voip-info.org/wiki-Asterisk+MeetMe2+Design it looks like a plan happened, but where is meetme2 at now? Things like recording a conference, allowing callers to adjust volume, allowing the conference to be locked, having the users name recorded before entering, and then played back to other callers on entrance etc etc. Are these things available now, or would they require development. Regards C File: ATT00036.txt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Signaling on PRI channels
On Sat, 12 Mar 2005, Laurent Tostain wrote: Hi, We did an interconnection with our carrier few days ago. But, I noticed that there was a signaling problem on our trnuk. In fact, Asterisk indicates that the call is answered when we received ALTERTING message from our carrier. This is PRI debug logs : [...] This is the zapata.conf file : [channels] [...] callprogress=yes I guess this could be the culprit. You do not want audio call progress analysis on a pri. Perhaps that is what is confusing Asterisk. Peter I changed callprogress to no in zapata.conf. And now, that works fine. Thanks peter ! Laurent ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with ability to dial out when a channel is used from an external equipment in a point to multi point configuration
I have grouped two capi controllers using * as the outgoing msn. when the first two channels are busy normally and I try to use a third channel the channels from the second controller are used. But when a channel is occupied by the capisuite fax and we need a third channel asterisk responds with every one is busy congested at this time. I think the same behavious would have happened if an external equipment was using the isdn channel in a point to multi point configuration. The same behavious happens when I unplug the cable from the ISDN controller 1. This means that when one ISDN line from the group of 4 lines isn't working then the pbx is unable to dial out. Who is going to fix it ? The version I use is CVS-HEAD-08/13/04-12:00:00-BRI-stuffed-0.1.0-RC4a ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Question on phones with asterisk
Hi, I've read the wiki... but would like some input from users here (not implying that wiki writers aren't users). I'm looking for a cheap (sub 60$) wired phone, or ATA device.. can anyone recommend one (or several), and possibly a source? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Does zapateller work in Australia?
as asked. -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime does not work yet, ...
You may not have most recent CVS. You should have this in your sip.conf: rtcachefriends=yes ; Cache realtime friends by adding them to the internal list ; just like friends added from the config file only on a ; as-needed basis. rtnoupdate=yes ; do not send the update request over realtime. rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule ; as if it had just registered when the registration expires ; the friend will vanish from the configuration until requested ; again. If set to an integer, friends expire ; within this number of seconds instead of the ; same as the registration interval NAT should be VARCHAR(5) If everything works fine when UA's are defined in sip.conf then there is most likely a db data issue. Try changing NAT as above. Be sure to use yes or no. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] playing invalid to an internal user
* is playing invalid message twice. I have: [internal] include = outgoing ; include the outgoing context include = voicemail ... include = invalid [invalid] exten = _.,1,Answer exten = _.,2,Playback(pbx-invalid) exten = _.,3,Hangup() asterisk is playing invalid message twice, WHY? -- Executing Answer(SIP/11-df84, ) in new stack -- Executing Playback(SIP/11-df84, pbx-invalid) in new stack -- Playing 'pbx-invalid' (language 'en') -- Executing Hangup(SIP/11-df84, ) in new stack == Spawn extension (internal, 6541, 3) exited non-zero on 'SIP/11-df84' -- Executing Answer(SIP/11-df84, ) in new stack -- Executing Playback(SIP/11-df84, pbx-invalid) in new stack -- Playing 'pbx-invalid' (language 'en') -- Executing Hangup(SIP/11-df84, ) in new stack == Spawn extension (internal, h, 3) exited non-zero on 'SIP/11-df84' -- #Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voice Based Bulletin Board.
First, I did my noob homework and found a thread where this had been discussed to some degree in a thread titled asterisk based bbs in 2004. I've got a question or two that that thread didn't address. I want to set up a voice based BBS. This will barely stray from the text based bulletin board paradigm we have now. Users will still go to a web page. There, they will still see threaded text based lists of topics and responses. The only differences are: 1. When a user clicks through text based link in a thread they will hear the users response instead of reading it; and 2. When creating a new topic or responding they would still enter the text based title, but instead of entering the text for their response they will enter their phone number and hit submit. Their phone rings and they record their post. Plusses: - Maintains many of the advantages of text based system including quick scanning of topics and a visual representation of the threads of discussion. - No more emoticons or cap locks. When I want to yell I'll yell! - I write some pretty long posts. Talking would save me some time...I think. Minuses: - Search is whacked beyond the posts' titles. - Your typical user will need a fatter pipe to read posts (but not to create them, a plus). - More storage needed. - More complicated setup. Now I am sure there is a way to get this done with asterisk and that is what I will eventually pursue. I was wondering though if I could create a fairly temporary solution that would work in a shared hosting (read rented server space) environment which is what I have now. It seems like it would be possible to hack this together with one of the existing VOIP service providers. I know they have voice mail forwarded as email. I could just have the mail forwarded to an account on my shared web server. They must be time stamped so I could use that to sort out which email goes with which post on the web server. The thing I don't know about the service providers is do they have an API to which an internet request can be made to from my web server to initiate a call that is supposed to be recorded. Do any of you know if any of them offer that? Are there asterisk hosting service providers that work along the same lines as web hosting companies? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hang on making progrogress passing when dialing out
I am getting the following on dial-out via Sipphone to a 1-800 number (numbers obscured): - == Spawn extension (macro-sipphone, s, 3) exited non-zero on 'SIP/eric-9546' in macro 'sipphone' == Spawn extension (default, 1747xxx, 1) exited non-zero on 'SIP/eric-9546' -- Executing Macro(SIP/eric-8e80, sipphone|1800xxx) in new stack -- Executing SetCallerID(SIP/eric-8e80, 1747xxx) in new stack -- Executing SetCIDName(SIP/eric-8e80, Eric Windisch) in new stack -- Executing Dial(SIP/eric-8e80, SIP/[EMAIL PROTECTED]||r) in new stack -- Called [EMAIL PROTECTED] -- SIP/proxy01.sipphone.com-8361 is making progress passing it to SIP/eric-8e80 == Spawn extension (macro-sipphone, s, 3) exited non-zero on 'SIP/eric-8e80' in macro 'sipphone' == Spawn extension (default, 1800xxx, 1) exited non-zero on 'SIP/eric-8e80' - System details: * I do not have any G729 licenses * The system is behind a two NAT routers with ports 5060 and 5061 forwarded on each. * externip and localnet are defined in sip.conf. * There is no special codec configuration in sip.conf (defaults) * SIP/eric is a Sipura 2100 with Preferred Codec = G711u, and Use Preferred Coded Only = no. * I am able to successfully make and receive calls through Zap/1, just not to sipphone. -- Eric Windisch [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
