Re: [asterisk-users] Pingable and Unreachable at the same time !
Asterisk doesn't use PING to check the STATUS, it uses a SIP OPTION message. Regards, Marc From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: mardi 17 février 2009 14:06 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Pingable and Unreachable at the same time ! Hi, Has anyone met something like this ? dialor*CLI sip show peers Name/username HostDyn Nat ACL Port Status 7541/7541 (Unspecified)D 0UNKNOWN 7540/7540 (Unspecified)D 0UNKNOWN 7534/7534 (Unspecified)D 0UNKNOWN 7533/7533 (Unspecified)D 0UNKNOWN 7531/7531 192.168.100.199 D 5060 OK (10 ms) 7530/7530 192.168.100.196 D 5060 UNREACHABLE patton/patton 192.168.100.52 D 5060 OK (33 ms) trunk/trunk4ipbx 192.168.64.25060 OK (1 ms) 8 sip peers [Monitored: 3 online, 5 offline Unmonitored: 0 online, 0 offline] dialor*CLI !ping 192.168.100.196 PING 192.168.100.196 (192.168.100.196) 56(84) bytes of data. 64 bytes from 192.168.100.196http://192.168.100.196: icmp_seq=1 ttl=64 time=0.334 ms 64 bytes from 192.168.100.196http://192.168.100.196: icmp_seq=2 ttl=64 time=0.305 ms 64 bytes from 192.168.100.196http://192.168.100.196: icmp_seq=3 ttl=64 time=0.305 ms Any explaination ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing with asterisk
The Attrafax software that was mentioned at the beginning of the thread does support Gateway mode. Regards, Marc -Original Message- Fabio Mosti wrote: 2009/2/16 Steve Underwood ste...@coppice.org: You don't indicate the kind of setup you are using. I use asterisk (Spandsp) with a IAX2 trunk (ethernet connection) to another asterisk (zap). client-asterisk (Spandsp)-asterisk (zap)-fax To quote the Mythbusters, there's your problem. Fax over IP = forget it unless the connection between your two Asterisk machines is some form of LAN connection. This *may* change a little when the T.38 support in Asterisk includes a gateway mode, which I don't believe it does yet. (IIRC 1.6 includes much better support for T.38, but I don't think it includes this kind of gateway yet - anyone care to correct me?) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Setting Request URI
Hello Douglas, I don't know if this is exactly what you need, but the fromdomain and fromuser in sip.conf (explained here: http://www.voip-info.org/wiki-Asterisk+config+sip.conf) change the From: header to [EMAIL PROTECTED] Regards, Marc Douglas Garstang wrote: Does anyone know how to set the request URI of SIP messages being sent from Asterisk to a peer? Asterisk always puts the IP address or hostname of the peer in the request URI. Eventhough Asterisk's SRV lookups are broken, I'd really like to put a domain name in the request URI (makes OpenSER routing easier and more logical). Anyone know how to do this? Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP response 484 Address Incomplete incorrectly handled
Hello, I saw that the error: SIP response 484 Address Incomplete is converted into DIALSTATUS = NOANSWER HANGUPCAUSE = 16 (NORMAL_CLEARING) shouldn't it be something like HANGUPCAUSE = 1 (UNALLOCATED) HANGUPCAUSE = 28 (INVALID_NUMBER_FORMAT) or another cause, other than NORMAL ??? Regards, Marc ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Register redirect
Hello, I would like to know if there is a way in IAX2 and SIP to tell a client to register at a different server. For example: Client tries to register at server B but server B answers with some sort of redirect to tell the client to register at server C. The client then tries to register with Server C. Best regards, Marc Storck. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Dial with 44 and +44 prefix
To bad that prefixes like +220 (Gambia), +230 (Mauritius), +240
(Equatorial Guinea), +250 (Rwanda), +260 (Zambia), +290 (Saint Helena),
+350 (Gibraltar), +370 (Lithuania), +380 (Ukraine), +420 (Czech
Republic), +500 (Falkland Island), +590 (Guadeloupe), +670 (Timor
Leste), +680 (Palau), +690 (Tokelau), +800 (IFPS), +850 (Northern
Korea), +870 (Inmarsat), +880 (Bangladesh) and +960 (Maldives) exist,
otherwise your example would have worked. But you may always include
these exceptions into your dialplan.
Regards,
Marc
Chris Bagnall wrote:
One further question, how can I set up a
line so that if 440 is dialled before a number the 0 is taken
out so only 44 is actually used?
exten = _440.,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:3})
You could probably do it by playing around with different offets as well:
exten = _440.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN::2}${EXTEN:3})
This would be more flexible if you wanted to do the same for different
country codes, for example:
exten = _NX0.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN::2}${EXTEN:3})
That would remove the zero from any 2-digit country code.
exten = _NXX0.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN::3}${EXTEN:4})
That'd do the same thing for a 3-digit country code.
Regards,
Chris
--
voipGATE.com
Support Team
___
--Bandwidth and Colocation sponsored by Easynews.com --
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] UK Pounds and pence prompt wanted
Yes there has just been a new release of Asterix (the Gaul has x at the end) . JP Carballo wrote: Obelix wrote: Is there a .gsm file for announcing UK pounds and pence after the credit remaining prompt, besides the dollar and cents file? /Obelix http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files+international I peeked into the archives from: http://www.desktop2door.com/asterisk/ and http://www.g7ltt.com/VoIP/vmfiles.html Found pound and pounds but no pence. I could have missed it though. You could also add your own voice to the UK male voice archive. That's what I did when I didn't find philippine(s).gsm My voice is nowhere near Allison's though. O.T. Is the Asterisk the Gaul comics still in circulation? It's been years since I read the series... -- voipGATE.com Support Team ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] One phone ringing, one phone flashing ?
this may work on Grandstream phones... set the ring tone to number 3 which is empty, so no tone and set the ring tone number 1 or 2 to ring on CallerID matching (e.g. everything starting with 678 will use ring tone 2) ... I never tested it, but the configuration shows the fields, so it may work. Regards, Marc Stefan-Michael. Guenther (in-put GbR) wrote: Hi, well, some clients have strange ideas and wishes (at least to my mind). Yesterday I gave a presentation about asterisk to a CEO. At the end he asked me whether asterisk is able to do the following: When a call for the CEO comes in, the calling number should be shown on the display of his phone and the phone of his secretary. The secretary's phones should ring, but at his phone only a light should flash. ;-)) No, turning off the sound isn't the solution. This restriction should e.g. only apply, when it is an external call, internal calls should result in ringing both phones. I'm not quite sure, whether this could be a feature of asterisk or the phone or both together. Does anything of you successfully set up something like this or could recommend a phone that would help/support it? Thanks a lot in advance, Stefan ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 -- voipGATE.com Support Team ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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: faxing to/from asterisk - new scripts
I would be interested as well... Why not post them somewhere? Regards, Marc [EMAIL PROTECTED] wrote: I'm game for using them /and testing them. Ben.. Roman: I created two bash scripts called Mail2Fax and Fax2Mail for use with the asterisk sever. They leverage the app_txfax and app_rxfax scripts, along with ast_fax. They make using these apps a lot easier, including being able to mail to [EMAIL PROTECTED] for outgoing faxes and then extracting phone numbers from the subject line! (Makes it easy to use with Sendmail without complex rules / virtual user tables). They also include error logs, parameter checking, etc. Let me know if you want them Michelle Dupuis Technical Support Specialist Oxford Consulting Group Ltd. Making IT work for your business... T: (519) 672-8238 E: mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] W: http://www.ocg.ca/ www.ocg.ca ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 Realtime Question
Hello, I tried to add the following SIP friend to SIP Realtime: [sip-friend23] type=friend host=12.13.14.15 context=acme disallow=all allow=ulaw allow=alaw accountcode=sip-friend23 But only calls to that SIP friend work, calls from that friend are instantly matched to the default context set in sip.conf. Can someone explain the right way to solve this situation? Best regards, Marc ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Switch between FXS ports
The Call Forward On Busy does cost YOU money each time you forward a call. Call hunting group is different from Call forwarding. In a hunt group you have 2 or more phone lines grouped together. When a call for a number associated with the group comes into the telco switch, the switch checks which lines inside the group are available, then the switch selects one of the available lines where it will send the call to. This selection is done using a predefined algorythm (random, round robin, ascending, descending,) Call hunting groups are also in most times used on a T1 PRI or E1 PRI. When a call comes in for a phonenumber associated with the T1/E1 only 1 channel will ring. Some telcos may charge additional fees to setup a call hunting group, but in cases you make a certain usage of Call forwarding, it may be less expensive to use a call hunting group. Best regards, Marc Rich Adamson wrote: I have two FXS port on my TDM card. channel 4 is attached with a telco line that I use frequently. And channel 3 have another telco line. but I dont publish that number to my friends. If I receive a call through channel 4, how can I handover that call to channel 3 ..so that I can keep channel 4 open for incoming call. Basically, you'd need to have the telco have the phone calls auto forwarded to the next available line. That's pretty common for them to do. That's exactly what I do with our business line. Call Forward on Busy is a common description for that telco service. (I simply forward that next call to an unlisted/unpublished number which also terminates in Asterisk.) ___ 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 -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] IT Service Providerhttp://www.msnetworks.lu 15, route d'Esch Phone: +352 2727 3030 L-4450 Belvaux Fax: +352 2727 3060 --- MS Networks powered service --- http://www.LuxAdmin.com Hosting and housing solutions --- ___ 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] Full T38 sip Faxing now Available
Do you want to share your knowledge how to get it work??? Regards, Marc Brian Chrystal wrote: i dont see what the big deal is. t38 works for me with * -Original Message- From: Brian Capouch [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 03, 2005 2:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Full T38 sip Faxing now Available Kanuri, Seshu (Company IT) wrote: Michael, Here are some of the reactions to your original post on the T38 FAX thingy: I have to add one thing to your list: what competent businessman would not realize that it's professional suicide to engage in so many questionable kinds of things on a public list like this? Michael: it was spam, for more than one reason: it was a commercial post and didn't belong here, you refused to follow up on repeated requests for more information, and still yet you haven't come clean with us just what kind of secret sauce you are touting--most likely there's not a bit of Open Source involved, and it may not even be related to Asterisk at all. You won't tell us. The whole deal smells to high heaven. Does it not give you pause to note that on this generally-friendly list, not a single person has ever come to your defense? You need to sign up for some getting along with others training, IMO. I can't imagine Sheltel going much of anywhere in this community, at least. Sorry. You just seem to always be embroiled in one controversy or another. That should give you pause. B. ___ 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 -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] IT Service Providerhttp://www.msnetworks.lu 15, route d'Esch Phone: +352 2727 3030 L-4450 Belvaux Fax: +352 2727 3060 --- MS Networks powered service --- http://www.LuxAdmin.com Hosting and housing solutions --- ___ 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] Full T38 sip Faxing now Available
The T38 service offered by this person has nothing to do with Asterisk, they want you to use their own system, and pay them for service. As far as I did understand, you need do install a custom firmware onto your ATA (only a limited number of ATAs are supported). Personally I don't think that the original post has anything to do with the purpose of this mailing list, as you cannot USE that service with ASTERISK. Regards, Marc Juan Jose Comellas wrote: Please send this information to me also. On Thu July 28 2005 01:03, Michael D Schelin wrote: Hello everybody, for all of you that have searched for a real fax solution, look no further. We now have T38 faxing. Please contact me for more information. Thanks Michael D. Schelin ShellTel 626-814-2354 ___ 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] Has Sixtel gone under?
I think so! Carlos Chavez wrote: I have been using Sixtel from the beginning of the year and service was getting worse and worse. Yesterday I tried to access the website to get the CDR and I got an error saying that the domain no longer exists. I checked the whois and it says that the domain is on hold. Have they finally folded? -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ 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 -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] IT Service Providerhttp://www.msnetworks.lu 15, route d'Esch Phone: +352 2727 3030 L-4450 Belvaux Fax: +352 2727 3060 --- MS Networks powered service --- http://www.LuxAdmin.com Hosting and housing solutions --- ___ 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] DID + 800 Providers
Thanks Michael, do they have an online ordering system, they don't seem to have a real website Regards, Marc Michael Graves wrote: On Sun, 24 Jul 2005 21:20:05 +0200, Marc Storck wrote: Hello, I'm looking for US DID and US50/CA 800# Providers. I found voiceconduits.com 8 month ago, there interface looks good, but there are still not live, I believe they won't be any time soon. I found sixtel, but order take eternities, they probably won't get my orders right any soon. So i'm looking for a good provider for DIDs and 800# from the US and CA, who offer online signup and ordering. The provisioning should be less than 12 hours, preferably instantly. If anybody knows or even uses such a provider, please leave me a note. I recommend www.clearpath1.com for 800 numbers. I've used them for a year and they've been absolutely reliable. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ 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 -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] IT Service Providerhttp://www.msnetworks.lu 15, route d'Esch Phone: +352 2727 3030 L-4450 Belvaux Fax: +352 2727 3060 --- MS Networks powered service --- http://www.LuxAdmin.com Hosting and housing solutions --- ___ 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] Why can't sip/200 call sip/202
: Digest username=200, realm=asterisk, algorithm=MD5, uri=sip:[EMAIL PROTECTED];user=phone, nonce=0c555366, response=ee6088fb4e50da5fe412913ae40dd45c Call-ID: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] CSeq: 45926 INVITE User-Agent: Grandstream GXP2000 1.0.1.9 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Type: application/sdp Content-Length: 258 v=0 o=200 8000 8001 IN IP4 192.168.0.3 s=SIP Call c=IN IP4 192.168.0.3 t=0 0 m=audio 5004 RTP/AVP 18 0 8 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 14 headers, 13 lines Using latest request as basis request Sending to 192.168.0.3 : 5060 (non-NAT) Found user '200' Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.3:5004 Found description format G729 Found description format PCMU Found description format PCMA Found description format telephone-event Capabilities: us - 0x10d (g723|ulaw|alaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Looking for 777 in default Reliably Transmitting (no NAT): SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKe570dfe4b8441304 From: Angus Comber sip:[EMAIL PROTECTED];user=phone;tag=a1afaf4fdb0ac845 To: sip:[EMAIL PROTECTED];user=phone;tag=as668982be Call-ID: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] CSeq: 45926 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.0.3:5060 Sip read: ACK sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKe570dfe4b8441304 From: Angus Comber sip:[EMAIL PROTECTED];user=phone;tag=a1afaf4fdb0ac845 To: sip:[EMAIL PROTECTED];user=phone;tag=as668982be Contact: sip:[EMAIL PROTECTED];user=phone Proxy-Authorization: Digest username=200, realm=asterisk, algorithm=MD5, uri=sip:[EMAIL PROTECTED];user=phone, nonce=0c555366, response=7fcb1024a81b3ea3bcc56baeca4bac3e Call-ID: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] CSeq: 45926 ACK User-Agent: Grandstream GXP2000 1.0.1.9 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Length: 0 12 headers, 0 lines Destroying call '[EMAIL PROTECTED]' mailto:'[EMAIL PROTECTED]' How can I troubleshoot? What should I be looking at? Angus ___ 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 -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] IT Service Providerhttp://www.msnetworks.lu 15, route d'Esch Phone: +352 2727 3030 L-4450 Belvaux Fax: +352 2727 3060 --- MS Networks powered service --- http://www.LuxAdmin.com Hosting and housing solutions --- ___ 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] DID + 800 Providers
Hello, I'm looking for US DID and US50/CA 800# Providers. I found voiceconduits.com 8 month ago, there interface looks good, but there are still not live, I believe they won't be any time soon. I found sixtel, but order take eternities, they probably won't get my orders right any soon. So i'm looking for a good provider for DIDs and 800# from the US and CA, who offer online signup and ordering. The provisioning should be less than 12 hours, preferably instantly. If anybody knows or even uses such a provider, please leave me a note. Many thanks, Marc ___ 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] Why can't sip/200 call sip/202
;
;[submenu]
;exten = s,1,Ringing ; Make them comfortable with 2 seconds of
ringback
;exten = s,2,Wait,2
;exten = s,3,Background(submenuopts) ; Thanks for calling the sales
department. Press 1 for steve, 2 for...
;exten = 1,1,Goto(default,steve,1)
;exten = 2,1,Goto(default,mark,2)
[default]
;
; By default we include the demo. In a production system, you
; probably don't want to have the demo there.
;
include = demo
;
; Extensions like the two below can be used for FWD, Nikotel, sipgate etc.
; Note that you must have a [sipprovider] section in sip.conf whereas
; the otherprovider.net example does not require such a peer definition
;
;exten = _41X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],,r)
;exten = _42X.,1,Dial(SIP/user:[EMAIL PROTECTED]:[EMAIL PROTECTED],30,rT)
; Real extensions would go here. Generally you want real extensions to
be 4 or 5
; digits long (although there is no such requirement) and start with a
single
; digit that is fairly large (like 6 or 7) so that you have plenty of
room to
; overlap extensions and menu options without conflict. You can alias
them with
; names, too and use global variables
;exten = 6245,hint,SIP/Grandstream1SIP/Xlite1 ; Channel hints for
presence
;exten = 6245,1,Dial(SIP/Grandstream1,20,rt) ; permit transfer
;exten = 6245,1,Dial(${HINT},20,rtT) ; Use hint as listed
;exten = 6361,1,Dial(IAX2/JaneDoe,,rm) ; ring without time limit
;exten = 6389,1,Dial(MGCP/aaln/[EMAIL PROTECTED])
;exten = 6394,1,Dial(Local/6275/n) ; this will dial ${MARK}
;exten = 6275,1,Macro(stdexten,6275,${MARK}) ; assuming ${MARK} is
something like Zap/2
;exten = mark,1,Goto(6275|1) ; alias mark to 6275
;exten = 6536,1,Macro(stdexten,6236,${WIL}) ; Ditto for wil
;exten = wil,1,Goto(6236|1)
;
; Some other handy things are an extension for checking voicemail via
; voicemailmain
;
;exten = 8500,1,VoicemailMain
;exten = 8500,2,Hangup
;
; Or a conference room (you'll need to edit meetme.conf to enable this
room)
;
;exten = 8600,1,Meetme(1234)
;
; Or playing an announcement to the called party, as soon it answers
;
;exten = 8700,1,Dial(${MARK},30,A(/path/to/my/announcemsg))
;
; For more information on applications, just type show applications at
your
; friendly Asterisk CLI prompt.
;
; 'show application command' will show details of how you
; use that particular application in this file, the dial plan.
;
- Original Message - From: dbruce [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, July 24, 2005 8:39 PM
Subject: Re: [Asterisk-Users] Why can't sip/200 call sip/202
Marc: My answer is not incorrect... it is incomplete.
The OP stipulated 2 extensions 200 and 202... and provided a sip debug
indicating a call from 200 to 777.
I pointed out the obvious.
If the OP is dialing 202 on the phone, and the phone is dialing 777,
then he
needs to look at the dialplan configuration of the phone. If he is
dialing
777 on the phone and expecting to reach 202, then he will need to have
translations in the asterisk dialplan. But, the question was what
should I
be looking at?... Using just the information provided, and the fact
that he
is new to asterisk... without any further information... the first
thing he
should be looking at is why the phone is trying to reach 777 when he
wants
to reach 202... Many new users do not realize the complexity of the SIP
protocol, and only really look at the trace in a general manner...
such as:
INVITE
407 Proxy Authentication Required
ACK
INVITE
404 Not Found
ACK
The idea was to provide a clue... not to provide a complete working
dialplan
and phone configuration. Providing new users with the complete
package is
a dis-service to them. They will only learn from thier mistakes and
experiences.. providing clues allows them to expand their experience and
build their confidence... It requires them to look at the details and
learn
to analyse them.
Regards,
Derek
- Original Message -
From: Marc Storck [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, July 24, 2005 12:53 PM
Subject: Re: [Asterisk-Users] Why can't sip/200 call sip/202
Derek: you reply is uncorrect. If Angus has the extension 777 in his
dialplan/extensions.conf which will dial 202. The name of the peer has
absolutely nothing to do with which number/name he would have to dial.
Without dialplan he will be unable to call any extension even 202, as
202 is only the name of the peer.
Angus: please paste your extensions.conf to pastebin.ca
Regards,
Marc
dbruce wrote:
It appears from the debug that extension 200 is trying to call 777,
not
202. Your Asterisk server can't find an extension 777 and returns 404
not found. That will explain why you can't call extension 777 from
extension 200. If you want to call extension 202, you will need to
dial
202 on extension 200, not 777.
Regards,
Derek
Re: [Asterisk-Users] Why can't sip/200 call sip/202
No please use ${EXTEN}, ${ARG1} is for macros.
And of course you will use the protocol in front of ${EXTEN}
So for SIP use:
exten = _2XX,1,Dial(SIP/${EXTEN},30)
and for IAX2 use:
exten = _2XX,1,Dial(IAX2/${EXTEN},30)
Regards,
Marc
Angus Comber wrote:
Would this do it:
exten = _2XX,1,Dial(${ARG1},30)
Then I would fallback to voicemail (or something else) after the 30
seconds?
Angus
- Original Message - From: Marc Storck
[EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, July 24, 2005 10:06 PM
Subject: Re: [Asterisk-Users] Why can't sip/200 call sip/202
Ok your extensions.conf doesn't mention anything about an
extension/number equal to 202 or 200. You must know that the name of a
SIP and IAX2 peer is only an address, you will have to assign a
number via extensions.conf to this address.
Have a look at www.voip-info.org and of course google.com to get to
know extensions.conf.
Regards,
Marc
Angus Comber wrote:
I think the 777 may be a bit of a Red Herring. I dialed 777 as a
test. I can't dial 202 from 200 if I actually dial 202!
My extensions.conf file:
;
; Static extension configuration file, used by
; the pbx_config module. This is where you configure all your
; inbound and outbound calls in Asterisk.
;
; This configuration file is reloaded
; - With the extensions reload command in the CLI
; - With the reload command (that reloads everything) in the CLI
;
; The General category is for certain variables.
;
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified. Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command 'save dialplan' too
;
writeprotect=no
; You can include other config files, use the #include command
(without the ';')
; Note that this is different from the include command that
includes contexts within
; other contexts. The #include command works in all asterisk
configuration files.
;#include filename.conf
; The Globals category contains global variables that can be
referenced
; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for
Environmental variable
; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
;
[globals]
CONSOLE=Console/dsp; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK=Zap/g2 ; Trunk interface
;
; Note the 'g2' in the TRUNK variable above. It specifies which group
(defined
; in zapata.conf) to dial, i.e. group 2, and how to choose a channel
to use in
; the specified group. The four possible options are:
;
; g: select the lowest-numbered non-busy Zap channel (aka. ascending
sequential hunt group).
; G: select the highest-numbered non-busy Zap channel (aka.
descending sequential hunt group).
; r: use a round-robin search, starting at the next highest channel
than last time (aka. ascending rotary hunt group).
; R: use a round-robin search, starting at the next lowest channel
than last time (aka. descending rotary hunt group).
;
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
;TRUNK=IAX2/user:[EMAIL PROTECTED]
;
; Any category other than General and Globals represent
; extension contexts, which are collections of extensions.
;
; Extension names may be numbers, letters, or combinations
; thereof. If an extension name is prefixed by a '_'
; character, it is interpreted as a pattern rather than a
; literal. In patterns, some characters have special meanings:
;
; X - any digit from 0-9
; Z - any digit from 1-9
; N - any digit from 2-9
; [1235-9] - any digit in the brackets (in this example,
1,2,3,5,6,7,8,9)
; . - wildcard, matches anything remaining (e.g. _9011. matches
; anything starting with 9011 excluding 9011 itself)
;
; For example the extension _NXX would match normal 7 digit
dialings,
; while _1NXXNXX would represent an area code plus phone number
; preceeded by a one.
;
; Each step of an extension is ordered by priority, which must
; always start with 1 to be considered a valid extension.
;
; Contexts contain several lines, one for each step of each
; extension, which can take one of two forms as listed below,
; with the first form being preferred. One may include another
; context in the current one as well, optionally with a
; date and time. Included contexts are included in the order
; they are listed.
;
;[context]
;exten = someexten,priority,application(arg1,arg2,...)
;exten = someexten,priority,application,arg1|arg2...
;
; Timing list for includes is
;
; time range|days of week|days of month|months
;
;include = daytime|9:00-17:00|mon-fri|*|*
;
; ignorepat can be used to instruct drivers to not cancel dialtone upon
; receipt of a particular pattern. The most commonly used example
Re: [Asterisk-Users] Connect 30 phone lines to asterisk how to
E1 or T1 card??? Regards, Marc Angel Diaz wrote: Hi, I have to connect 30 phone lines to my asterisk server, can somebody help on how I have to do it ? I have a TDM405P and one TDM400P with 4 FXO ports. Do I have to use 8 TDM400P ? Or, is there another way to do it ? Thanks, Angel. ___ 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 -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] IT Service Providerhttp://www.msnetworks.lu 15, route d'Esch Phone: +352 2727 3030 L-4450 Belvaux Fax: +352 2727 3060 --- MS Networks powered service --- http://www.LuxAdmin.com Hosting and housing solutions --- ___ 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] blindtransfers with IAX
Hello,
I use the ${BLINDTARNSFER} variable for transfers from SIP accounts, but
this variable seems to be unavailable for IAX channels. Is this supposed
to be this way, is there another variable???
Many thanks for your help,
Marc
___
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] Quotation request: 12 KHz signal generation for billing purposes.
In most european countries you will need 16kHz. It would be interesting for us as well, but I think it may just work with alaw/ulaw. Regards, Marc Cenk Yabas wrote: Could anyone quote a price for the following project. We should be able to generate a specific (say 12Khz) signal at certain intervals (calculated using a price/rate table on a mySQL database) DURING an ongoing conversation. The conversation is to be marked (start and end) with specific signals as well. This is a requirement for special hotel applications where a device counts the signals to calculate a price for the ACTUAL (after successful connect) conversation. We believe, this task may require some source code modifications on Asterisk. Thanks, Cenk Yabas. ___ 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 -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] IT Service Providerhttp://www.msnetworks.lu 15, route d'Esch Phone: +352 2727 3030 L-4450 Belvaux Fax: +352 2727 3060 --- MS Networks powered service --- http://www.LuxAdmin.com Hosting and housing solutions --- ___ 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] IAX aproprietary protocol
IAX is an abbreviation for Inter Asterisk Exchange. So IAX was a proprietary protocol for interconnecting Asterisk servers, it was only used with 2 asterisk servers. IAX has always been open for the community. So some may say it's proprietary, while it is open. At the current time, the IAX protocol is not only used in asterisk, but also in some softphone clients and other software, but you still need asterisk as one of the partners in a client-server relation. So wether you can still call it proprietary is up to you. Regards, Marc Joseph wrote: Can anybody explain me why IAX is called proprietary protocol? In some places IAX is refereed as open protocol. How can proprietary protocol be open protocol? -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] IT Service Providerhttp://www.msnetworks.lu 15, route d'Esch Phone: +352 2727 3030 L-4450 Belvaux Fax: +352 2727 3060 --- MS Networks powered service --- http://www.LuxAdmin.com Hosting and housing solutions --- smime.p7s Description: S/MIME Cryptographic Signature ___ 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] IAX aproprietary protocol
How can proprietary protocol be open protocol? Proprietary means it came from a proprietor - Digium in this case. This is a completely unrelated issue to whether it is open. Marketing departments try to confuse the issues. :-) So if the protocol is not encumbered by any patent or copyright (only by missing documentation) it shouldn't be referred as proprietary as it only confusing and hurts the cause. Laziness of others (to write the documentation or implement the protocol) should not qualify to label the IAX2 as proprietary. Even WIKI is confusing the cause calling it proprietary without any valid reason :-) http://www.voip-info.org/tiki-index.php?page=Asterisk%20protocols I think that should be corrected! Documentation is here: http://www.cornfed.com/iax.pdf -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] IT Service Providerhttp://www.msnetworks.lu 15, route d'Esch Phone: +352 2727 3030 L-4450 Belvaux Fax: +352 2727 3060 --- MS Networks powered service --- http://www.LuxAdmin.com Hosting and housing solutions --- smime.p7s Description: S/MIME Cryptographic Signature ___ 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] Shanghai or Bangalore DIDs
Hello, does someone offer DIDs from the areas of shanghai and/or bangalore. Many thanks, Marc -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] IT Service Providerhttp://www.msnetworks.lu 15, route d'Esch Phone: +352 2727 3030 L-4450 Belvaux Fax: +352 2727 3060 --- MS Networks powered service --- http://www.LuxAdmin.com Hosting and housing solutions --- smime.p7s Description: S/MIME Cryptographic Signature ___ 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] Shanghai or Bangalore DIDs
I'm also looking for numbers from HongKong, Taiwan, Japan and Singapore So if someone has some DIDs from this areas, I'm very interested to get one or another from those DIDs. Best Regards, Marc Marc Storck wrote: Hello, does someone offer DIDs from the areas of shanghai and/or bangalore. Many thanks, Marc ___ 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 -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] IT Service Providerhttp://www.msnetworks.lu 15, route d'Esch Phone: +352 2727 3030 L-4450 Belvaux Fax: +352 2727 3060 --- MS Networks powered service --- http://www.LuxAdmin.com Hosting and housing solutions --- smime.p7s Description: S/MIME Cryptographic Signature ___ 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] pridialplan/TON question
Anyway, if I set TON to unknown, I have to send the number according to the PSTN dialing plan (00 for int, 0 for national, ...). Thus, if the PBX does not use UNKNOWN, I have to translate the numbers out of their original TON to ton=unknown. Therefore, I need to process the incoming TON. How do I handle this? You have two options: 1) Use the CALLINGTON variable in the dialplan. This is only for the calling party number, not the called party number. 2) Use the internationalprefix, nationalprefix, localprefix etc settings in the zapata.conf file. I _think_ this will affect both the interpretation of calling and called party and possibly also the TON of the called number for outgoing links. I am nut sure under which circumstances these variables are applied. Isdn handling in Asterisk tends to be these kinds of hacks. I have a Digium E100P card, with an EuroISDN PRI E1. On incoming calls the CALLINGTON variable is empty. I have the latest stable version of asterisk. Do I have to use another variable or is the TON only support in CVS? Marc -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] IT Service Providerhttp://www.msnetworks.lu 15, route d'Esch Phone: +352 2727 3030 L-4450 Belvaux Fax: +352 2727 3060 --- MS Networks powered service --- http://www.LuxAdmin.com Hosting and housing solutions --- smime.p7s Description: S/MIME Cryptographic Signature ___ 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] How to prevent native bridging between SIP channels
add canreinvite=no to the sip user definition blocks for the SIP provider and for the SIP ATA. Regards, Marc Wolf N. Paul wrote: Hello, how can I prevent Asterisk from trying to create a native bridge between an incoming call from a SIP provider and an extension attached to a SIP ATA? My Asterisk is behind a firewall, and the native bridge invariably fails. Thanks in advance for any suggestion! (I DID search the list archives for native bridge and found one similar query without any replies). Regards, Wolf Paul ___ 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 -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] IT Service Providerhttp://www.msnetworks.lu 15, route d'Esch Phone: +352 2727 3030 L-4450 Belvaux Fax: +352 2727 3060 --- MS Networks powered service --- http://www.LuxAdmin.com Hosting and housing solutions --- ___ 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 Restart after crash
After applying the change to the init script, it seems to restart the asterisk processes which get killed, but do you have a functional system with this? Our Testsystem spits out some '100% CPU-Loaded mpg123 processes' and asterisk was somehow dead. Did I miss something? Yes I found the same thing. Looks like either safe_asterisk needs to kill any mpg123 processes before restarting asterisk or the patch to only run an mp3 player when necessary that exists IIRC becomes the standard. Can someone point me in the right direction so I can find that patch? Regards, Marc ___ 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] can't make my PRI dial out
Mark, that's what the command pri debug span 1 does, produce a lot of output so you can see what is received and what is sent. Maybe you can paste the output to pastebin.ca and tell us the link. Regards, Marc Mark Phillips wrote: Nothing happens. I get the same (non)error. I get plenty of output when receiving a call however. Mark Andrew Kohlsmith wrote: On April 22, 2005 10:41 am, Robert Webb wrote: Your zapata.conf should look like this: language=en context=default switchtype=4ess pridialplan=unknown signalling=pri_cpe echocancel=yes group=1 channel=1-23 You need to move the echocancel and the group above the channel line. The channel line definitions must be above and not below. You're right, but that's not his problem. Cause code 0 is no cause code at all; I'd turn on pri debug span 1 output and see what's coming up there. -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 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] Junghans QuadBRI and fax detection
I use DIDs for incoming faxes as well, but we have several users with
combined Telephone-Fax-Hardware. As humans make errors and are very
lazy, these users don't want to dial another prefix when they send a fax.
This is what I try to do:
exten = _XX.,1,SetVar(NUMBER=${EXTEN}) ;save the number they dialled
exten = _XX.,2,Answer() ; answer to check if it is a fax
exten = _XX.,3,Dial(IAX2/[EMAIL PROTECTED]/${NUMBER}) ; if it's NOT a
fax dial via IAX
exten = fax,1,Dial(Zap/g1/${NUMBER}) ; if it IS a fax, dial via ISDN
In my situation it does not work, no call is forwarded to the fax extension.
Many thanks,
Marc
Kristof Hardy wrote:
Eugenio De Vena wrote:
I have QuadBRI and asterisk 1.0.6 bristuffed but fax reception works
ugly.
My faxes are missing many
rasters and even sending does not work well. Can you tell me with
version of
asterisk , spandsp, app_sndfax etc
you use to have a good result?
I have used bristuff-0.2.0-RC7k (as on junghanns.net ).
This downloads v1.0.6 of asterisk, zaptel and libpri.
I'm not doing fax-detection, I'm using DID to dedicate 1 number to fax
receiving. Os is Debian Sarge.
Let me know if it works out now ;)
___
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] Junghans QuadBRI and fax detection
Hello, does the Junghans QuadBRI Card and qozap module support Fax detection? I want to use fax detection using the Answer() command and the 'fax' extension. I used the example from the wiki, but I had no success so far. Can someone please share his/her experiences/knowledge?? Many thanks, Marc ___ 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] chat line
Setup an IVR to take care of the menu you described, and use different meet-me rooms per destinations. Marc James Taylor wrote: Yes, the meetme can be part of it. I was thinking more of a classified ad chat line, you know the male-female thing: ...If you are a man looking for a woman, press one... ...If you are a woman looking for a man, press two... ...If you are not sure, press three... ...If you don't care, press four... ...If you are a dog looking for his master, press five... I'll share my notes if someone wants to team up and work on this. James On Sun, 13 Mar 2005 13:31:43 GMT, Iqbal [EMAIL PROTECTED] wrote: am working on it for a client, and yes as Steven said, I think the meetme will do itnow just to figure out the billing part :-) On this note as anyone thought of premium line SIP addresses...I know this may sounds strange, and SIP--SIP is normally free (this I feel will change once voice hits critical mass) but just a thought Iqbal On 3/13/2005, Steven Critchfield [EMAIL PROTECTED] wrote: On Sat, 2005-03-12 at 21:35 -0600, James Taylor wrote: Anyone done a chat line app? Any reason why the meetme app doesn't fullfuill your needs or did you not bother to look? -- Steven Critchfield [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 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 -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] IT Service Providerhttp://www.msnetworks.lu 15, route d'Esch Phone: +352 2727 3030 L-4450 Belvaux Fax: +352 2727 3060 --- MS Networks powered service --- http://www.LuxAdmin.com Hosting and housing solutions --- ___ 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 signalling and RTP to different servers
Hello, we're in process of testing an interconnection with a trans-european carrier. But the carrier wants the SIP signalling to server 1 and the RTP stream to server 2. How do I configure asterisk to work with that type of installation. It seems they are using NexTone as SIP Signaling and RTP servers. Can someone help me??? Regards, Marc -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] IT Service Providerhttp://www.msnetworks.lu 15, route d'Esch Phone: +352 2727 3030 L-4450 Belvaux Fax: +352 2727 3060 --- MS Networks powered service --- http://www.LuxAdmin.com Hosting and housing solutions --- ___ 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] where is voice conduits
According to http://www.itu.int/ITU-T/inr/forms/files/Applications-E-164.pdf Page 3 the 882 99 has been assigned to Telenor (http://www.telenor.com). So e164.org may have a problem with that prefix, if the 882 99 is ever used by Telenor. Regards, Marc ross jones wrote: on 2/28/05 09:49, Andrew Thompson at [EMAIL PROTECTED] wrote: There was a thread a month or two ago on here about voiceconduits. The general gist was they are not yet open for public business. Are there any voice conduits customers out there? if not, maybe I ought to just walk away. -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] IT Service Providerhttp://www.msnetworks.lu 15, route d'Esch Phone: +352 2727 3030 L-4450 Belvaux Fax: +352 2727 3060 --- MS Networks powered service --- http://www.LuxAdmin.com Hosting and housing solutions --- ___ 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] where is voice conduits
Sorry for replying into the wrong thread. Regards, Marc Marc Storck wrote: According to http://www.itu.int/ITU-T/inr/forms/files/Applications-E-164.pdf Page 3 the 882 99 has been assigned to Telenor (http://www.telenor.com). So e164.org may have a problem with that prefix, if the 882 99 is ever used by Telenor. Regards, Marc ross jones wrote: on 2/28/05 09:49, Andrew Thompson at [EMAIL PROTECTED] wrote: There was a thread a month or two ago on here about voiceconduits. The general gist was they are not yet open for public business. Are there any voice conduits customers out there? if not, maybe I ought to just walk away. -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] IT Service Providerhttp://www.msnetworks.lu 15, route d'Esch Phone: +352 2727 3030 L-4450 Belvaux Fax: +352 2727 3060 --- MS Networks powered service --- http://www.LuxAdmin.com Hosting and housing solutions --- ___ 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] where is voice conduits
Oups I shouldn't have left that much voice messages those last weeks ;-) I once got to talk with someone from voiceconduits via AIM, but that's all, no reply to emails and voicemail! Marc ross jones wrote: Does any one know what happened with voice conduits? I have been trying to reach them for nearly three weeks now. Their voice mail boxes are full and writing email to them does not get any returns. Thoughts or sightings are appreciated. -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] IT Service Providerhttp://www.msnetworks.lu 15, route d'Esch Phone: +352 2727 3030 L-4450 Belvaux Fax: +352 2727 3060 --- MS Networks powered service --- http://www.LuxAdmin.com Hosting and housing solutions --- ___ 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] Type of Number
Hello,
how can I read the PRI type of number:
[ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan
E.164/E.163) (1)
Presentation: Presentation allowed of network provided number (3)
'061706161' ]
(in this case TON = 2)
Does a variable like ${TON} exist??? Or how can i read that number?
If this would have to be implemented I'm willing to fund a bounty!
Regards,
Marc
--
CTOMarc Storck
MS Networks SA [EMAIL PROTECTED]
Internet Service Provider http://www.msnetworks.lu
15, route d'Esch Phone: +352 2727 3030
L-4544 Belvaux Fax: +352 2727 3060
- MS Networks powered service -
http://www.Gateway.lu Your gateway to the net
Advantages of ADSL solutions by MS Networks:
- price: cheap and clear
- products: proven quality
- support: friendly and helpful
---
___
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] VoiceConduits is a scam
if they are really from sri lanka, than I can understand why they don't understand, as you may know Sri Lanka was hit by an Earth Quake followed by a Zunami. I for my part filled a complaint at paypal, I got instantly refunded, either Paypal knows more than we all, or I cannot explain it... Marc Kristian Kielhofner wrote: Tim Mattison wrote: I've paid them, tried to provision numbers, e-mailed support, instant messaged support, and got nowhere. I highly recommend everyone stays away from this provider. Total scam, I have not signed up with them, but from the looks of it they are a complete scam: 1) Every number on the site has been disconnected. 2) The Whois record indicates a contact tel. number in Sri Lanka. 3) Too good to be true. Someone should try to take their site down (the legal ways, of course...). -- Kristian Kielhofner ___ 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 -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] Internet Service Provider http://www.luxadmin.org 15, route d'Esch Phone: +352 2727 3030 L-4544 Belvaux Fax: +352 2727 3060 - MS Networks powered service - http://www.Gateway.lu Your gateway to the net Advantages of ADSL solutions by LuxAdmin: - price: cheap and clear - products: proven quality - support: friendly and helpful --- ___ 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] VoiceConduits - Notice, Apology, and Clarification
Hello, what made me think most of a scam was the fact that nobody responded to my support request, nobody sent me a notice that the service is not yet actively marketed, there was/is no notice on the website. If only 5 users where able to sign up and I'm one of them, then it shouldn't have been that difficult to contact those users and tell them what's up with voiceconduits. I would be happy to get the ordered TollFree numbers online... I hope they were not part of the deal with that company just passing away Althought I didn't get a call, please accept my apologies, I'll be more than happy to use your services as soon as they are available, or even as test-user (but please tell me so ;-) ) Thank you for your messages to the list, Marc Kristian Kielhofner wrote: [EMAIL PROTECTED] wrote: Hello, This is David Deutsch, and Im the owner of VoiceConduits. There seems to be some confusion related to our company, regarding the past few posts. VoiceConduits is currently NOT open for public business, we have never to date advertised or attempted to attract business. It appears that a few people heard about our company via a mention in a SineApps article and found our beta system that is under development. We apologize that a few people managed to sign up via this interface, and we will happily refund anyone who did so immediately, additionally we will supply them with free credit to be used once we are in fact live. It was certainly never our intention to defraud individuals of the asterisk or voip community, our understanding is that only 5 people have managed to signup thru this automated system, and we will be contacting each of them individually to insure they are refunded and happy with the resolution. Thank you, David Deutsch, President Tris Telecommunications, LLC (800) 547-4057 x1001 David, Thank you for calling me and clearing things up. It seems that once you have VoiceConduits up and running it will be an excellent service. I really do look forward to doing business with VoiceConduits. For the record: 1) The Sri Lanka number was a cell phone in India, where David was helping with the tsunami. 2) The Wyoming \ Nevada inc. thing, was done to minimize personal liability, as in Delaware. Why I don't necessarily agree with it, I can certainly understand why you would want to do it... In the world of spam and shoddy internet transactions, it is not that hard to at least make your company LOOK legitimate. This includes valid contact numbers, good communication, accurate WHOIS records, etc. I can appreciate that you were not ready for business, but it is not that hard to block the sign-up portion of your site until you are ready. Hopefully this whole thread can be a lesson to someone looking to setup a business and avoid this kind of thing. Brian, David, anyone else, my sincerest apologies. -- Kristian Kielhofner ___ 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 -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] Internet Service Provider http://www.luxadmin.org 15, route d'Esch Phone: +352 2727 3030 L-4544 Belvaux Fax: +352 2727 3060 - MS Networks powered service - http://www.Gateway.lu Your gateway to the net Advantages of ADSL solutions by LuxAdmin: - price: cheap and clear - products: proven quality - support: friendly and helpful --- ___ 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] Music instead of Tunes
Hello, more and more operators in Europe offer music instead of ring tunes. E.g. instead of the 400 Hz or whatever tunes, the caller will hear J-Lo, or Mozart Currently I will have to answer the line to do that. Is there a way to do this with asterisk? Regards, Marc -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] Internet Service Provider http://www.luxadmin.org 15, route d'Esch Phone: +352 2727 3030 L-4544 Belvaux Fax: +352 2727 3060 - MS Networks powered service - http://www.Gateway.lu Your gateway to the net Advantages of ADSL solutions by LuxAdmin: - price: cheap and clear - products: proven quality - support: friendly and helpful --- ___ 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 behind IX66
Hi, I seem to remember from the TelAppliant HowTo PDF, that you actually have to create an individual entry in sip.conf for each Number, just as you do for each SipGate number. Regards, Marc Steve Beaumont wrote: Hi, I have a problem with asterisk behind an IX66 router. Outgoing calls are OK at this time, I say this because I have occasions when outgoing calls fail. For incoming calls I consistantly get the following error message:- Dec 26 15:44:10 NOTICE[23533]: chan_sip.c:7183 handle_request: Failed to authenticate user 01256removed sip:01256removed@217.10.79.218;tag=as18ff3a97 This error messages occurs when processing an incoming call from voiptalk (telappliant). I also experiane a similar problem with sipgate and possibly FWD. FWD fallsback to FWD voice mail. All peers show as registered when using 'sip show registry'. I can't track down what is going wrong. If I configrue an entry in sip.conf for the specific number, in this case '01256,removed' the call is processed correctly. However, this approach is not possible as I would have to enter every PSTN number into sip.conf, even I'm not that mad :-) The IX66 is running v3.11. he sip configuration is basic i.e. left as basic proxy. I've various settings and alsways seem to get the best results if I leave the IX66 at the default settings. I'm not sure if this is the correct approach. Any cluses where ot look would be apreciated. Driving me further round the bend :-) config snips follow:- sip.conf [general] context=inbound register=username:password@gw3.voiptalk.org/extn to forward to ; Note gw3.voiptalk.org advised by telappliant [telappliant] type=friend username=username secret=password context=outbound-to-voiptalk host=gw3.voiptalk.org dtmfmode=info insecure=very canreinvite=no Best regards - Steve Beaumont Email: steveb (at) asbc.demon.co.uk mailto:[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 -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] Internet Service Provider http://www.luxadmin.org 15, route d'Esch Phone: +352 2727 3030 L-4544 Belvaux Fax: +352 2727 3060 - MS Networks powered service - http://www.Gateway.lu Your gateway to the net Advantages of ADSL solutions by LuxAdmin: - price: cheap and clear - products: proven quality - support: friendly and helpful --- ___ 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] gateway.lu
Hello, we do not use MD5 secrets and for that customer account we have the following codecs activated: GSM ULAW ALAW ILBC which should be available and working with any asterisk version! We also see the asterisk of the customer as registered. The customer's * is behind NAT, our server is on a public IP. So there must be something else causing that error. Regards, Marc Race Vanderdecken wrote: 1. Sometimes you get a 403 if the CODECs do not match Do a sip debug by starting Asterisk with: Asterisk -dvgc Then at the command line interace type: CLI sip debug Look for a CODEC match line. 2. Maybe the are asking for MD5 security Lock for the word nonce= in the sip debug log If they are sending a nonce then you have to do MD5 stuff. Race The Tyrant Van der Decken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jorge Mendoza Sent: 21 December 2004 18:43 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] gateway.lu Is somebody using this service? I have problems connecting my Asterisk server with SIP. I tried different configurations and always receive 403 Forbidden message. My SIP config is: [general] register = 1234:[EMAIL PROTECTED] [voip1-out] type=peer secret=4321 username=1234 host=voip1.gateway.lu fromuser=1234 nat=yes My Asterisk server is nated. Any clue? Thanks Jorge Mendoza ___ 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 -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] Internet Service Provider http://www.luxadmin.org 15, route d'Esch Phone: +352 2727 3030 L-4544 Belvaux Fax: +352 2727 3060 - MS Networks powered service - http://www.Gateway.lu Your gateway to the net Advantages of ADSL solutions by LuxAdmin: - price: cheap and clear - products: proven quality - support: friendly and helpful --- ___ 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] Dialing out to 2 clients simultaneously
Hello, this is not possible, you will have to solve this via the dialplan using parallel ringing or queues. Regards, Marc [EMAIL PROTECTED] wrote: Hi When I register a SIP or IAX client to asterisk and I dial to it from another UA then there is no problem at all But, when I register two or more clients to the SAME peer (with the same user/pass) and I call to this peer.. Then only the UA which registered the last will ring.. Others don't ring... What can I do about this?? I would like to register for example 10 UA's to the same peer and want them all to ring at the same time without having to set up different usernames and passwords for all these ua's and having to make difficult dialplans Is this possible? Am I doing something wrong or is this behaviour by design? Regards, Niels ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] Internet Service Provider http://www.luxadmin.org 15, route d'Esch Phone: +352 2727 3030 L-4544 Belvaux Fax: +352 2727 3060 -- LuxAdmin powered service --- http://www.Gateway.lu Your gateway to the net Advantages of ADSL solutions by LuxAdmin: - price: cheap and clear - products: proven quality - support: friendly and helpful --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Ethernet Channel Bank (Comming Soon to a NOC Near You!)
Will you have a Channel Bank for E1, many E1s instead of the FXS/FXO ports?? Marc Christopher Dobbs wrote: My company has started development on a Ethernet based channel bank. Here are the (current) spec's - 10/100 Ethernet Port - Up to 96 FXS/FXO ports (Thats 4 DS1's for the math impaired) - Serial Console - TDMoE - IAX2 - EETP (A protocol that we have designed for IP Telephony) We have just started prototyping this device, so... -- Christopher Dobbs ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] Internet Service Provider http://www.luxadmin.org 15, route d'Esch Phone: +352 2727 3030 L-4544 Belvaux Fax: +352 2727 3060 -- LuxAdmin powered service --- http://www.Gateway.lu Your gateway to the net Advantages of ADSL solutions by LuxAdmin: - price: cheap and clear - products: proven quality - support: friendly and helpful --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DUNDi performance
Hello, I have a weird problem. My * server, a Pentium Celeron 1200 with 512 MB Ram and a Digium E100P card, is performing very well for IAX2, SIP and ZAP communication. There is no delay in transcoding, no packet loss etc etc. Now I added DUNDi, and I added +/- 8 peers in the dundi-test context and 1 peer in the GPA-bound e164 context. My server shows all but 1 peer as OK. DUNDi Ping times are between 20 and 200 ms. The Problem is, that no server but one can get a stable connection via DUNDi to my server. DUNDi ping times for my server are between 3000 an 7000 ms. Most servers have qualify of 2000ms, some even 500ms, so my server is quite always UNREACHABLE for those peers. When I activate DUNDi DEBUG, I can see that incoming DUNDi packets do take all long time before they show up in the * console, and they always show up with a whole bunch of others (filling 2-3 screens). But then the debug output stops in the middle of 1 debug packet, to continue over 20 seconds later (if it continues). The actual CPU load is load average: 0.00, 0.00, 0.00. I cannot find the problem, maybe someone over can help me! Thanks, Marc -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] Internet Service Provider http://www.luxadmin.org 15, route d'Esch Phone: +352 2727 3030 L-4544 Belvaux Fax: +352 2727 3060 -- LuxAdmin powered service --- http://www.Gateway.lu Your gateway to the net Advantages of ADSL solutions by LuxAdmin: - price: cheap and clear - products: proven quality - support: friendly and helpful --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Base Number and DIDs
Hello, one of the numbers where historically configured to act the following way: 123456: Ring All Desks 123456-1: Ring Desk 1 123456-2: Ring Desk 2 ... (I think you get the idea) Configuring asterisk to do the same isn't that hard, but I now have one problem, with users calling that number from PSTN. Those particular users go off-hook and start dialing the number. The ZAP Channel claims a match at 123456 it isn't waiting for an evt. digit that may follow. The calling user does have a problem when he dials the number and goes off-hook (the phone will dial the number at high-speed), the same is true if the users uses the redial button. So my question is, how can make asterisk or the ZAP channel wait a little bit longer before he claims a match against a number in the dialplan... Thanks, Marc -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] Internet Service Provider http://www.luxadmin.org 15, route d'Esch Phone: +352 2727 3030 L-4544 Belvaux Fax: +352 2727 3060 -- LuxAdmin powered service --- http://www.Gateway.lu Your gateway to the net Advantages of ADSL solutions by LuxAdmin: - price: cheap and clear - products: proven quality - support: friendly and helpful --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] SCRIPT: Fax Remvoal Please Call: 1-800...
exten = 666,1,Dial(1800number,180,D(1yourfaxnumber1) exten = 666,2,Hangup i don't know if you can add another priority after hangup. Regards, Marc P.S.: try 'show application dial' for details Joseph wrote: At time to time I receive some junk faxes from some advertising companies that play smart and don't provide any TSI number so I can not bock them by the number in Hylafax. Despite calling their Fax Removal Service 1-800-... number several time they refuse to obey my request. So I would like to setup a small script or context loop in extension.conf if possible and maybe run it overnight; maybe I get their attention if nothing else works! Does anybody have any idea how to do it? In extension.conf it would be something like: exten = 666,1,Dial(1800number) ; How to go next priority after 10sec.? exten = 666,2,Wait 10 ;wait for voice message to finish, and wait for tone exten = 666,3,Dial(my-fax-number) ;after about 10sec. exten = 666,4,Dial(1) ;to confirm selection exten = 666,5,Hangup exten = 666,6,Goto(s,1) Any improvements are welcome. -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] Internet Service Provider http://www.luxadmin.org 15, route d'Esch Phone: +352 2727 3030 L-4544 Belvaux Fax: +352 2727 3060 -- LuxAdmin powered service --- http://www.Gateway.lu Your gateway to the net Advantages of ADSL solutions by LuxAdmin: - price: cheap and clear - products: proven quality - support: friendly and helpful --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] What about a higher level configuration language
Why VMxyz, does every line end up at the VM when it's busy or unavailable or unregistered btw we could then also add a rule for the case the user agent has registered with * (bristuff addon n+201) Best Regards, Marc Asterisk wrote: I pray for an end to the priorities as well. The +101 could be easily solved by a default label, or an option to the dial for example: exten = _7XX,1,Dial(yada,10) exten = _7XX,2,Voicemail(unavail) exten = _7XX,3,Hangup exten = _7XX,102,Voicemail(Busy) could be: exten = Dial:_7XX,Dial(yada,10) exten = Hangup:_7XX,Hangup exten = VMUnavail:_7XX,Voicemail(unavail) exten = VMBusy_7XX,Voicemail(Busy) in other words, the dial automatically looks for VMUnavail if not answered, or VMBusy if the line is busy or exten = StartPlan:Dial:_7XX,Dial(yada,10,BeforeAnswer=AA,AfterAnswer=ZZ,Busy=XX,NoAnswer=YY) exten = ZZ:_7XX,Hangup exten = XX:_7XX,Voicemail(unavail) exten = YY:_7XX,Voicemail(Busy) There must be fat better ways of expressing my thoughts, but it's late on Sunday :) Julian - Original Message - From: Brian Capouch [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, September 26, 2004 8:41 PM Subject: Re: [Asterisk-Users] What about a higher level configuration language Dinesh Nair wrote: On 27/09/2004 00:50 Jay Milk said the following: Eliminating the need to specify (and keep track of) priorities would make changes to extensions.conf much easier to implement. or perhaps allow non-consecutive priorities. After this topic was discussed a bit at the developer's confab, I got to thinking about what a great feature that would be. Renumbering priorities is a sadly common task for me in my somewhat chaotic config environment, and having a way to sneak in actions in between existing ones would be a major win. Of course, the problem of the hard-coded priority + 101 situation is problematical. I say we think through what the perfect world would look like in this respect and then see how hard it would be to implement. . . B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] Internet Service Provider http://www.luxadmin.org 15, route d'Esch Phone: +352 2727 3030 L-4544 Belvaux Fax: +352 2727 3060 -- LuxAdmin powered service --- http://www.Gateway.lu Your gateway to the net Advantages of ADSL solutions by LuxAdmin: - price: cheap and clear - products: proven quality - support: friendly and helpful --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Question about the 'fax' extension
If you use extension dedicated to fax, then you don't need to use the fax extenstion, but just call the rxfax application directly as you would call the answer application exten = 123456,1,rxfax(...) But of course you may just use different fax extensions for different contexts. Regards, Marc [EMAIL PROTECTED] wrote: I was looking at the wiki on 'Asterisk as a voice/fax switch' And was wondering if the extension 'fax' is global to extensions.conf Or just to the context it is in? The reason I ask, is that my PRI might have 5 channels that will be scrictly Fax, and to be functional, I need multiple 'fax' extensions in my various Contexts. Hope that makes sense, Paul Seniuk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] Internet Service Provider http://www.luxadmin.org 15, route d'Esch Phone: +352 2727 3030 L-4544 Belvaux Fax: +352 2727 3060 -- LuxAdmin powered service --- http://www.Gateway.lu Your gateway to the net Advantages of ADSL solutions by LuxAdmin: - price: cheap and clear - products: proven quality - support: friendly and helpful --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] 1 extension entry for multiple purposes?
this works great for me, i use callerid= like this:
callerid=Marc Storck 35227273033
Matthew Boehm wrote:
OK. Here is the caveat I've found. The phones, in sip.conf, all have a
callerid= line because if they don't when they call someone the caller id
shows up ONLY as their extension.
For instance, my extension is 3044. When I call my cell, all it says is
Missed call from 3044.
The only way I found to fix this was to add that callerid= into the sip.conf
But since I have done that, what you have suggested below won't work.
Should I have the callerid set somewhere else?
Matthew
- Original Message -
From: Wiley E. Siler [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Monday, September 20, 2004 4:14 PM
Subject: RE: [Asterisk-Users] 1 extension entry for multiple purposes?
Here you go... No extension required
From extensions.conf
;--
; VOICEMAIL ENTRY INTO SYSTEM
;--
exten = 8,1,Answer
exten = 8,2,Wait(1)
exten = 8,3,VoicemailMain(${CALLERIDNUM})
exten = 8,4,Hangup
Still want the old way of enter your number then PIN...
exten = 81,1,VoicemailMain2()
exten = 81,2,Hangup
-Original Message-
From: Matthew Boehm [mailto:[EMAIL PROTECTED]
Sent: Monday, September 20, 2004 2:07 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] 1 extension entry for multiple purposes?
Hey gang,
There must be any easy solution for this but my mind is frazzled on
compiling 2.4 with RTC as module. Bleh.
Currently extension 9000 is our VoicemailMain(@company) line. Some
employee's are complaining that the old system was better because you
didn't have to enter your mailbox number and that instead the old system
took you right to it.
I figured there was something similar so that I don't have to have 200
extra extensions.conf lines just for VoicemailMain(exten@company).
Basically I want something like this: exten =
9000,1,VoicemailMain([EMAIL PROTECTED])
so that way all it asks for is their password.
Any ideas..?
Thanks,
Matthew
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
The information transmitted is intended only for the person or entity to
which it is addressed and may contain confidential and/or privileged
material. Any review, retransmission, dissemination or other use of, or
taking of any action in reliance upon, this information by persons or
entities other than the intended recipient is prohibited. If you received
this in error, please contact the sender and delete the material from any
computer
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
CTOMarc Storck
MS Networks SA [EMAIL PROTECTED]
Internet Service Provider http://www.luxadmin.org
15, route d'Esch Phone: +352 2727 3030
L-4544 Belvaux Fax: +352 2727 3060
-- LuxAdmin powered service ---
http://www.Gateway.lu Your gateway to the net
Advantages of ADSL solutions by LuxAdmin:
- price: cheap and clear
- products: proven quality
- support: friendly and helpful
---
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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] Question about the 'fax' extension
Answer() will jump to the fax extension in the same context just automatically... Marc [EMAIL PROTECTED] wrote: What happens if I want it to work over the same DiD though? Does Answer() take care of this? How do I jump to the fax extension if it detects a faxtone? Paul Seniuk -Original Message- From: mstorck [mailto:[EMAIL PROTECTED] Sent: September 20, 2004 3:51 PM To: asterisk-users Subject: Re: [Asterisk-Users] Question about the 'fax' extension If you use extension dedicated to fax, then you don't need to use the fax extenstion, but just call the rxfax application directly as you would call the answer application exten = 123456,1,rxfax(...) But of course you may just use different fax extensions for different contexts. Regards, Marc [EMAIL PROTECTED] wrote: I was looking at the wiki on 'Asterisk as a voice/fax switch' And was wondering if the extension 'fax' is global to extensions.conf Or just to the context it is in? The reason I ask, is that my PRI might have 5 channels that will be scrictly Fax, and to be functional, I need multiple 'fax' extensions in my various Contexts. Hope that makes sense, Paul Seniuk -- -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] Internet Service Provider http://www.luxadmin.org 15, route d'Esch Phone: +352 2727 3030 L-4544 Belvaux Fax: +352 2727 3060 -- LuxAdmin powered service --- http://www.Gateway.lu Your gateway to the net Advantages of ADSL solutions by LuxAdmin: - price: cheap and clear - products: proven quality - support: friendly and helpful --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Sending Caller ID info in MD/USA
Hmmm BTN, may be I should tell that to my E1 provider, they use my CallerID as the BTN, additionnaly whenever I don't provide a number which matches the one configured on my PRI the set a default CallerID!!! So that's also the reason why they cannot provide me with such a service like send-any-callerid-you-like!!! Too bad!!! Marc Jon Radel wrote: [EMAIL PROTECTED] wrote: On Wednesday 15 September 2004 01:02 am, Thomas Gallaway wrote: It entirely depends on how that carrier deals with caller ID. Usually you would not be able to set your own, as it's done by their equipment. It would just ignore yours. On a PRI I've never had a carrier, ILEC, CLEC, not-LEC-at-all, *not* expect me to provide the number I want to use for the source of the call. After all, many applications require many DIDs to be in service on a single PRI, and the carrier has no idea which one the call is from unless your PBX, soft switch, etc., tells them. Now, some of them will block anything other than the DIDs you've paid them for on that PRI, but that's a different story. (I say the above fully expecting that someone can tell me about an exception.) Being this is usually only handed over to them by other TELCOs, it's pretty much a new issue, and not something they are really prepared for. And all those little PBXs sitting in closets all over the land PRIs are not exactly a new product for end-user use. Everything could start falling to pieces if that does not work as all the billing is based on identifying who is calling whom. Which is exactly why any carrier with their act even mildly together also tracks the BTN (billing telephone number) of the PRI, based on the circuit your call comes in on, no matter what you claim as the source of the call. --Jon Radel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] Internet Service Provider http://www.luxadmin.org 15, route d'Esch Phone: +352 2727 3030 L-4544 Belvaux Fax: +352 2727 3060 -- LuxAdmin powered service --- http://www.Gateway.lu Your gateway to the net Advantages of ADSL solutions by LuxAdmin: - price: cheap and clear - products: proven quality - support: friendly and helpful --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 hangup
Hello, I have an E1 connected to an * server, which takes incoming calls and verifies the existance of the called number in our internal E164 tree. Now there is a number that exists on one of the servers, but the phone has registered itself, so the dial plan executes an hangup. This hangup however is not transmitted to the E1, the calling party hears no dial tone, but also no hangup or congestion or anything else just nothing. This is the console log of the E1 server passing the call via e164 resolution an SIP to another server: -- Executing EnumLookup(Zap/1-1, 1234567890) in new stack -- Accepting call from '' to '1234567890' on channel 0/1, span 1 -- Executing SetAccount(Zap/1-1, 0.00) in new stack -- Executing Dial(Zap/1-1, SIP/[EMAIL PROTECTED]|180|r) in new stack -- parse_srv: SRV mapped to host host.domain.tld, port 5060 -- Called [EMAIL PROTECTED] -- SIP/host.domain.tld-b41e answered Zap/1-1 == Spawn extension (PRI, 1234567890, 3) exited non-zero on 'Zap/1-1' -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (PRI, h, 1) exited non-zero on 'Zap/1-1' cdr_odbc: Query Successful! -- Hungup 'Zap/1-1' On the SIP server i see the following: -- Executing Macro(SIP/1.2.3.4-08442ea8, callextmbx|1234567890|SIP/1234567890|abc) in new stack -- Executing SetMusicOnHold(SIP/1.2.3.4-08442ea8, abc) in new stack -- Executing Dial(SIP/1.2.3.4-08442ea8, SIP/1234567890|60|r) in new stack Sep 15 01:51:16 NOTICE[1147739056]: app_dial.c:739 dial_exec Unable to create channel of type 'SIP' == Everyone is busy/congested at this time -- Executing GotoIf(SIP/1.2.3.4-08442ea8, 1?3:104) in new stack -- Goto (macro-callextmbx,s,3) -- Executing VoiceMail(SIP/1.2.3.4-08442ea8, u1234567890) in new stack Sep 15 01:51:16 WARNING[1147739056]: app_voicemail.c:1962 leave_voicemail: No entry in voicemail config file for '1234567890' -- Executing VoiceMail(SIP/1.2.3.4-08442ea8, b1234567890) in new stack Sep 15 01:51:16 WARNING[1147739056]: app_voicemail.c:1962 leave_voicemail: No entry in voicemail config file for '1234567890' -- Executing Hangup(SIP/1.2.3.4-08442ea8, ) in new stack == Spawn extension (macro-callextmbx, s, 105) exited non-zero on 'SIP/1.2.3.4-08442ea8' in macro 'callextmbx' == Spawn extension (CONTEXT, 1234567890, 1) exited non-zero on 'SIP/1.2.3.4-08442ea8' -- Executing Hangup(SIP/1.2.3.4-08442ea8, ) in new stack == Spawn extension (CONTEXT, h, 1) exited non-zero on 'SIP/1.2.3.4-08442ea8' cdr_odbc: Query Successful! When I call from another VoIP device it just works fine! I hope someone has some help! ;-) Regards, Marc -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] Internet Service Provider http://www.luxadmin.org 15, route d'Esch Phone: +352 2727 3030 L-4544 Belvaux Fax: +352 2727 3060 -- LuxAdmin powered service --- http://www.Gateway.lu Your gateway to the net Advantages of ADSL solutions by LuxAdmin: - price: cheap and clear - products: proven quality - support: friendly and helpful --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] TDMoE questions
Hello, thanks for the answers!!! You mentionned to use the switch command. I read about it in the WIKI, but I couldn't find enought information to understand what it is actually doing. Can someone point me to the right direction? Marc Steven Critchfield wrote: On Sat, 2004-09-11 at 21:41, Marc Storck wrote: Hello, I want to link several * boxes together. Some of them are dedicated as user servers (SIP and IAX clients connect to them) and some are used as PRI servers (where the PRIs are hooked onto). I think TDMoE is the only channel type where you can group different Interfaces into a single group. E.g. for using Dial(ZAP/g1/12345), I think you cannot group different IAX accounts and use them via Dial(IAX/g1/12345). Or am I wrong?? IAX with groups doesn't make sense. IAX being a network protocol is not physical port limited like PSTN hardware. Your trick here is to understand that you can dial via IAX from one machine to another and the second machine then takes the incoming call and does it's own Dial(Zap/g1/12345). Or with the use of a switch command, the IAX connections to the other side is implied and the remote side says it can complete the call so the user machine says, okay, do it. TDMoE has a limitation of X channels per link, and some people have noted troubles when trying to use more than one TDMoE circuit. IAX has no trouble talking to mulitple places and multiple calls. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] TDMoE questions
is there any in-depth information available about the switch command??? Marc Steven Critchfield wrote: On Sun, 2004-09-12 at 06:42, Marc Storck wrote: Hello, thanks for the answers!!! You mentionned to use the switch command. I read about it in the WIKI, but I couldn't find enought information to understand what it is actually doing. Can someone point me to the right direction? The switch command is helpful in tieing multiple machines together. Switch allows you to have one asterisk machine ask another if it can complete a call. Think of it a bit like a remote include. Basically from a context on machine A, you set up a switch statement to machine B in a specific context. Whenever machine B says it can complete the call, you end up with an implied IAX call to machine B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDMoE questions
Hello, I want to link several * boxes together. Some of them are dedicated as user servers (SIP and IAX clients connect to them) and some are used as PRI servers (where the PRIs are hooked onto). I think TDMoE is the only channel type where you can group different Interfaces into a single group. E.g. for using Dial(ZAP/g1/12345), I think you cannot group different IAX accounts and use them via Dial(IAX/g1/12345). Or am I wrong?? So I looked at the WIKI and it shows an example using em signalling. What other signallings are supported by TDMoE? How many TDMoE trunks with 30 channels each may I run on a 100 Mbit/s LAN dedicated to TDMoe??? Regards, Marc ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Net2Phone, Asterisk, and 404 Not Found
did you try to add canreinvite=yes to [net2phone3] ?? Marc [EMAIL PROTECTED] wrote: Hi! Net2Phone is getting a common SIP status code, 404 Not Found, when trying to place a call to our Asterisk server. We're hoping someone on the list can shed some light on why this is happening. We can process a call from Asterisk to Net2Phone without any problems. Net2Phone sends the INVITE but immediately gets the 404 Not Found. The To: field of the INVITE contains the E.164 formatted number with a plus + sign before the 11 digits and we were thinking that the presence of that plus sign had something to do with the 404 problem. But I guess the plus sign is part of the SIP standard. I don't think we've seen the INVITE but I'll dig further on that. Has anyone connected Asterisk to a different SIP proxy and used SIP to communicate between the two? Can anyone further explain why our Asterisk is not replying to Net2Phone's INVITE? Here is the entry from our sip.conf file: [net2phone3] context = n2p-in host=/Net2Phone's IP/ /Address/ disallow=g723.1 allow=g729 type=friend dtmfmode=rfc2833 Thanks in advance! chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] New G.729 codec and VLANS
You can re-register the codecs one time using other NICS. after that one time you need to contact Digium to be able to re-register, but the process is very easy! At 21:33 18.07.2004, you wrote: Anton Tinchev wrote: The readme says that the license uses all network cards MACS What happens when VLANS are added or removed? Is it safe? Also, in this day of motherboard-integrated NICs (even two or three), what will happen if the mobo dies and has to be replaced? ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] IRC
2 different things, you should be able to join a channel even if nickserv didn't authenticate you yet!!! but this is OT ;-))) Marc At 20:12 19.06.2004, you wrote: Steve Underwood wrote: Hi, I figured it out. Most IRC channels requiring some authentication give a minute's latitude to allow for slow response from nickserv. It seems #asterisk is not doing that. You really must wait for nickserv to say you are registered before you issue a /join #asterisk. You can't login to an ssh session until you are authenticated, why would IRC be any different? Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] e164.org
configure your asterisk to use e164.org and make use of EnumLookup then try to call +352 818 595, if your call goes to [EMAIL PROTECTED] then you can call me for free over the net! Marc At 03:33 23.05.2004, you wrote: Dean Collins wrote: Tony, as per you inference that e164 are up to something shady, you should talk to one of the founders Duane, he currently has about 5 open If it's the same duane who runs cacert he probably means well... however having read the site I'm still not sure whether i'd use it myself (it means trusting an external database to produce a least cost route.. I'm just not that trusting). Tony -- Te audire no possum. Musa sapientum fixa est in aure. Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13 Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] VoicePulse SIP
asterisk only supports IAX2, SIP and TEL, it will only use IAX2 and SIP entries however so it is used to route via the Net if it cannot find a route via the Net or the link isn't working it will go to the next priority in your dialplan and do whatever you want, it doesn't re-configure your dialplan or route preferences let's say it's a bypass to IAX and SIP providers as it will tell you the username and server where users may be reached directly!!! Marc At 23:50 22.05.2004, you wrote: Andres wrote: [EMAIL PROTECTED] wrote: Which providers give you a jitter buffer? In Europe: VoipTalk and Magrathea. In the US: Iconnecthere. I am sure there are more. Clearpath gives jitter buffer as well. http://www.clearpath1.com/ John ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Fehler beim starten...
try to ask in english you may get an answer a whole lot faster Regards, Marc - Original Message - From: Administrator [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, May 06, 2004 6:11 PM Subject: [Asterisk-Users] Fehler beim starten... Hallo, nachdem mir bis jetzt noch niemand geantwortet hat nochmal meine frage: wenn ich asterisk starte bekomme ich folgende fehlermeldung: [app_capiCD.so]May 6 00:38:23 WARNING[16384]: loader.c:240 ast_load_resource: /usr/lib/asterisk/modules/app_capiCD.so: undefined symbol: ast_capi_MessageNumber May 6 00:38:23 WARNING[16384]: loader.c:408 load_modules: Loading module app_capiCD.so failed! Ich habe SUSE 9 installiert, eine Fritzcard... usw. vielleicht kann mir ja jemand von euch helfen! Vielen Dank! mfg Markus Dohnal ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
