Re: [asterisk-users] Inserting a wait in a sip dial
12 jan 2010 kl. 20.56 skrev David Gibbons: snip 'w' is really only supported on channels where digit-by-digit dialing is the norm, which generally means analog trunks (or digital trunks using CAS signaling). /snip Thanks Kevin, that's what I figured (though not quite so concisely)... Going foward, is there any way to programmatically inject DTMF tones into an already-bridged channel? So: 1. dial 12345 2. connect SIP provider to * extension 3. wait 2 seconds programmatically 3. inject 567 DTMF tones into channel to signal remote PBX of extension to dial I'm hoping there's another way to skin this cat. From show application dial D([called][:calling]) - Send the specified DTMF strings *after* the called party has answered, but before the call gets bridged. The 'called' DTMF string is sent to the called party, and the 'calling' DTMF string is sent to the calling party. Both parameters can be used alone. /O -- _ -- 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] Major benefits when upgrade from Asterisk 1.4 to Asterisk 1.6?
13 jan 2010 kl. 06.56 skrev hadi motamedi: Dear All I have Asterisk 1.4 installed on my Debian server . I am considering upgrading my Asterisk to the latest version (1.6) . Can you please let me know what are the major benefits when upgrading from Asterisk 1.4 to Asterisk 1.6 ? Please observe that there is no 1.6 version. Previous to 1.6.0, there was 1.0 and 1.2 and 1.4. Then the release policy changed and we had 1.6.0 and 1.6.1 and now (latest) 1.6.2. These three 1.6.x are all very, very different. Also note that none of these are LTS releases, something that was recently introduced. LTS is Long Term Support. 1.4 is classified as LTS, and the next LTS will be version 1.8 - yes, we're correcting the mistake and going back to the old way of numbering releases. Personally, I see the 1.6.x releases as very experimental and don't recommend them for production use. In regards to changes, there has been a massive amount of changes, especially work done by the Digium dev team to rebuild the internal structure of Asterisk to support massive scalability and improve stability of Asterisk. The major new feature is of course faxing, that was introduced in 1.6.0 and has been improved in every release. Please download the new version and read the documentation that covers the CHANGES as well as instructions for upgrading your product. As always, there's no reason to upgrade if there are no features you need. 1.4 is still a supported release. Best regards, /Olle -- _ -- 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] Inserting a wait in a sip dial
12 jan 2010 kl. 19.47 skrev Danny Nicholas: Looking out for shots back on this, but Dial(SIP/X,w1234) should produce a 1/2 second delay before dialing, ww1234 a 1 second delay, etc. Try it with 2 or 3 w's instead of 1... I have no solution, but can only say this: a 'w' in a SIP dialstring doesn't produce any wait protocol-wise. SIP is all enbloc signalling. The gateway from SIP to PSTN might have an implementation of old hayes-like commands and support w for inserting wait periods, but you will have to check the documentation for that gateway. /O -- _ -- 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] Major benefits when upgrade from Asterisk 1.4 to Asterisk 1.6?
13 jan 2010 kl. 06.56 skrev hadi motamedi: Dear All I have Asterisk 1.4 installed on my Debian server . I am considering upgrading my Asterisk to the latest version (1.6) . Can you please let me know what are the major benefits when upgrading from Asterisk 1.4 to Asterisk 1.6 ? Please observe that there is no 1.6 version. Previous to 1.6.0, there was 1.0 and 1.2 and 1.4. Then the release policy changed and we had 1.6.0 and 1.6.1 and now (latest) 1.6.2. These three 1.6.x are all very, very different. Also note that none of these are LTS releases, something that was recently introduced. LTS is Long Term Support. 1.4 is classified as LTS, and the next LTS will be version 1.8 - yes, we're correcting the mistake and going back to the old way of numbering releases. Personally, I see the 1.6.x releases as very experimental and don't recommend them for production use. In regards to changes, there has been a massive amount of changes, especially work done by the Digium dev team to rebuild the internal structure of Asterisk to support massive scalability and improve stability of Asterisk. The major new feature is of course faxing, that was introduced in 1.6.0 and has been improved in every release. Please download the new version and read the documentation that covers the CHANGES as well as instructions for upgrading your product. As always, there's no reason to upgrade if there are no features you need. 1.4 is still a supported release. Best regards, /Olle -- _ -- 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] Major benefits when upgrade from Asterisk 1.4 to Asterisk 1.6?
13 jan 2010 kl. 06.56 skrev hadi motamedi: Dear All I have Asterisk 1.4 installed on my Debian server . I am considering upgrading my Asterisk to the latest version (1.6) . Can you please let me know what are the major benefits when upgrading from Asterisk 1.4 to Asterisk 1.6 ? Please observe that there is no 1.6 version. Previous to 1.6.0, there was 1.0 and 1.2 and 1.4. Then the release policy changed and we had 1.6.0 and 1.6.1 and now (latest) 1.6.2. These three 1.6.x are all very, very different. Also note that none of these are LTS releases, something that was recently introduced. LTS is Long Term Support. 1.4 is classified as LTS, and the next LTS will be version 1.8 - yes, we're correcting the mistake and going back to the old way of numbering releases. Personally, I see the 1.6.x releases as very experimental and don't recommend them for production use. In regards to changes, there has been a massive amount of changes, especially work done by the Digium dev team to rebuild the internal structure of Asterisk to support massive scalability and improve stability of Asterisk. The major new feature is of course faxing, that was introduced in 1.6.0 and has been improved in every release. Please download the new version and read the documentation that covers the CHANGES as well as instructions for upgrading your product. As always, there's no reason to upgrade if there are no features you need. 1.4 is still a supported release. Best regards, /Olle -- _ -- 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] Major benefits when upgrade from Asterisk 1.4 to Asterisk 1.6?
On Wed, Jan 13, 2010 at 7:36 AM, Olle E. Johansson o...@edvina.net wrote: 13 jan 2010 kl. 06.56 skrev hadi motamedi: Dear All I have Asterisk 1.4 installed on my Debian server . I am considering upgrading my Asterisk to the latest version (1.6) . Can you please let me know what are the major benefits when upgrading from Asterisk 1.4 to Asterisk 1.6 ? Please observe that there is no 1.6 version. Previous to 1.6.0, there was 1.0 and 1.2 and 1.4. Then the release policy changed and we had 1.6.0 and 1.6.1 and now (latest) 1.6.2. These three 1.6.x are all very, very different. Also note that none of these are LTS releases, something that was recently introduced. LTS is Long Term Support. 1.4 is classified as LTS, and the next LTS will be version 1.8 - yes, we're correcting the mistake and going back to the old way of numbering releases. Personally, I see the 1.6.x releases as very experimental and don't recommend them for production use. In regards to changes, there has been a massive amount of changes, especially work done by the Digium dev team to rebuild the internal structure of Asterisk to support massive scalability and improve stability of Asterisk. The major new feature is of course faxing, that was introduced in 1.6.0 and has been improved in every release. Please download the new version and read the documentation that covers the CHANGES as well as instructions for upgrading your product. As always, there's no reason to upgrade if there are no features you need. 1.4 is still a supported release. Best regards, /Olle -- _ -- 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 Thank you for your reply . I downloaded the Asterisk-1.6.2.0 and left my libpri-1.4 and zaptel-1.4 as they are . After the installation , according to you , I just have the fax feature that is being added . Can you please confirm if nothing wrong in my case? -- _ -- 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] Major benefits when upgrade from Asterisk 1.4 to Asterisk 1.6?
My apologies for the multiple copies. Had issues with a mailserver that somehow wasn't talking to DNS properly. Now fixed. It behaved like Asterisk does sometimes, very poor when it can't connect to DNS. Had power outage yesterday and I think that started it all... Meanwhile, I tried to retransmit to find the issue, as I noticed that my mails did not reach the list. Guess what, they all did in the end... ;-) /O -- _ -- 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] Major benefits when upgrade from Asterisk 1.4 to Asterisk 1.6?
On Wed, Jan 13, 2010 at 8:27 AM, Olle E. Johansson o...@edvina.net wrote: My apologies for the multiple copies. Had issues with a mailserver that somehow wasn't talking to DNS properly. Now fixed. It behaved like Asterisk does sometimes, very poor when it can't connect to DNS. Had power outage yesterday and I think that started it all... Meanwhile, I tried to retransmit to find the issue, as I noticed that my mails did not reach the list. Guess what, they all did in the end... ;-) /O -- _ -- 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 Thank you . Receiving your multiple replies is no problem at all . Looking forward your reply on upgrading to Asterisk-1.6.2.0 -- _ -- 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] Major benefits when upgrade from Asterisk 1.4 to Asterisk 1.6?
13 jan 2010 kl. 09.26 skrev hadi motamedi: On Wed, Jan 13, 2010 at 7:36 AM, Olle E. Johansson o...@edvina.net wrote: 13 jan 2010 kl. 06.56 skrev hadi motamedi: Dear All I have Asterisk 1.4 installed on my Debian server . I am considering upgrading my Asterisk to the latest version (1.6) . Can you please let me know what are the major benefits when upgrading from Asterisk 1.4 to Asterisk 1.6 ? Please observe that there is no 1.6 version. Previous to 1.6.0, there was 1.0 and 1.2 and 1.4. Then the release policy changed and we had 1.6.0 and 1.6.1 and now (latest) 1.6.2. These three 1.6.x are all very, very different. Also note that none of these are LTS releases, something that was recently introduced. LTS is Long Term Support. 1.4 is classified as LTS, and the next LTS will be version 1.8 - yes, we're correcting the mistake and going back to the old way of numbering releases. Personally, I see the 1.6.x releases as very experimental and don't recommend them for production use. In regards to changes, there has been a massive amount of changes, especially work done by the Digium dev team to rebuild the internal structure of Asterisk to support massive scalability and improve stability of Asterisk. The major new feature is of course faxing, that was introduced in 1.6.0 and has been improved in every release. Please download the new version and read the documentation that covers the CHANGES as well as instructions for upgrading your product. As always, there's no reason to upgrade if there are no features you need. 1.4 is still a supported release. Best regards, /Olle Thank you for your reply . I downloaded the Asterisk-1.6.2.0 and left my libpri-1.4 and zaptel-1.4 as they are . After the installation , according to you , I just have the fax feature that is being added . Can you please confirm if nothing wrong in my case? I'm sorry, I don't understand you. Please check the documentation to find all new features added, the CHANGES file is a good start. If you update to latest Asterisk, I think you should update libpri and change zaptel to Dahdi to get access to the latest features of all packages. /O -- _ -- 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] Hardware issue, was Using HASH() and REALTIME_HASH()
Le 12/01/2010 16:35, Tilghman Lesher a écrit : On Tuesday 12 January 2010 04:44:36 Benoit wrote: I just experienced another problem however i have two rnis cards, one b410p and one te220, while the later works prefectly i can't really make the first one work, using DAHDI or mISDN under asterisk 1.6. If you're having trouble with any Digium hardware, the best thing to do is to call Digium support and get your free installation support provided with our hardware. Hi, I didn't think of this, since it looked like more of an asterisk problem (asterisk 1.4/misdn = ok asterisk 1.6/misdn = fail, asterisk 1.6/dahdi = fail). Audio (both way) is working (voicemail/playback), but it fail when Dial'ing a device. Looks like a problem with signalling ... But anyway i just opened a support case, thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is roundrobin and rrmemory the same meaning?
Yes it's actually quite simple to do. If you want, the free version of QueueMetrics is able to do that from the Agent's page. l. 2010/1/13 Zhang Shukun bit...@gmail.com 2010/1/12 Lenz Emilitri lenz.lo...@gmail.com: You can list phones directly as static members of the queue. i know i can configure the queue.conf and agents.conf to add queue name and queue members by hand. Could i use functions to create queue name and add queue members dynamiclly. because i want to create a call center use asterisk, which users can register their own call number on the web site. also they can add several service phone numbers along with a fix extension (like:1), the phone numbers are customer service number, when it's customer dial the call number and press extension 1, one member should answer the caller. so, when configured on the web. like: extension 1:12345, 12346,12347,12348,12349 when finished the data above should stored in the database, when user call in and press 1. i should create a queue and add 12345, 12346,12347,12348,12349 to the queue. is this possible? -- Loway - home of QueueMetrics - http://queuemetrics.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Major benefits when upgrade from Asterisk 1.4 to Asterisk 1.6?
On Wed, Jan 13, 2010 at 8:49 AM, Olle E. Johansson o...@edvina.net wrote: 13 jan 2010 kl. 09.26 skrev hadi motamedi: On Wed, Jan 13, 2010 at 7:36 AM, Olle E. Johansson o...@edvina.net wrote: 13 jan 2010 kl. 06.56 skrev hadi motamedi: Dear All I have Asterisk 1.4 installed on my Debian server . I am considering upgrading my Asterisk to the latest version (1.6) . Can you please let me know what are the major benefits when upgrading from Asterisk 1.4 to Asterisk 1.6 ? Please observe that there is no 1.6 version. Previous to 1.6.0, there was 1.0 and 1.2 and 1.4. Then the release policy changed and we had 1.6.0 and 1.6.1 and now (latest) 1.6.2. These three 1.6.x are all very, very different. Also note that none of these are LTS releases, something that was recently introduced. LTS is Long Term Support. 1.4 is classified as LTS, and the next LTS will be version 1.8 - yes, we're correcting the mistake and going back to the old way of numbering releases. Personally, I see the 1.6.x releases as very experimental and don't recommend them for production use. In regards to changes, there has been a massive amount of changes, especially work done by the Digium dev team to rebuild the internal structure of Asterisk to support massive scalability and improve stability of Asterisk. The major new feature is of course faxing, that was introduced in 1.6.0 and has been improved in every release. Please download the new version and read the documentation that covers the CHANGES as well as instructions for upgrading your product. As always, there's no reason to upgrade if there are no features you need. 1.4 is still a supported release. Best regards, /Olle Thank you for your reply . I downloaded the Asterisk-1.6.2.0 and left my libpri-1.4 and zaptel-1.4 as they are . After the installation , according to you , I just have the fax feature that is being added . Can you please confirm if nothing wrong in my case? I'm sorry, I don't understand you. Please check the documentation to find all new features added, the CHANGES file is a good start. If you update to latest Asterisk, I think you should update libpri and change zaptel to Dahdi to get access to the latest features of all packages. /O -- _ -- 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 Thank you very much for your reply . I am using from Asterisk:The future of telephony book to install Asterisk . For the new version , according to you , I am using from Asterisk-1.6.2.0 , Libpri 1.4.10.2 , and Dahdi 2.2.0.2 . But for Dahdi installation , the mentioned book does not have any section . Can you please confirm if the Dahdi installation is the same as Libpri or not? -- _ -- 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] Xorcom 32 channel FXS gateway
On Tue, Jan 12, 2010 at 05:53:02PM -0600, Carlos Chavez wrote: On Tue, 2010-01-12 at 18:05 -0500, C F wrote: Anyone on the list ever used it? I'm trying to quote a system with 192 analog ports, one of the options are the Xorcom 32 channel FXS USB Channel Banks. Any input would be appreciated. I have used Astribanks for a while now and they are usually very stable. The only thing to worry about is that if you change the order the units are connected to the USB ports then you will have a mess on your hands. This has been mostly addressed in DAHDI 2.2 with the xpp_order file. Once you're satisfied with the order of Astribanks, run: dahdi_genconf xpporder Which will generate /etc/dahdi/xpp_order . Astribanks listed in this file get registered first by dahdi_registration . So the bottom line is that you can make the order of registration more predicatable (either by label or by connector - see comments in that file). -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about SIP registration
I reply to your question below 1) I don't have a secret for that peer. 2) Obviously, the solution is to make the 'host' field static (in my scenario, because the port is non-standard 5080, so no standard endpoint SIP can register with that IPaddress:port) or specify a secret with 'host=dynamic'. The question I made was a little different: I'm wondering why an external SIP endpoint, which is trying to register on eth0 85.X.Y.Z network, is indeed seen by Asterisk as registered with address 1.1.1.1 (the eth1 IP addresses of the PC). I try to explain better: usually, SIP endpoint with IP address X.Y.Z.T which has registered itself on Asterisk (for example with user 200) is seen as following (sip show peers) 200/200 X.Y.Z.T5060OK(xx ms) So the CLI shows the *endpoint's* IP address. Instead, in my scenario, I see a row like this: 999/999 1.1.1.15060UNREACHABLE (1) And 1.1.1.1 is the eth1 IP address of the PC where Asterisk is installed on. But I haven't any endpoint SIP onto that PC which is trying to register, while I can see one of them OUTSIDE my network (i.e. in the Big Internet) that is trying to register as 999: in fact, if in [999] SIP account I put 'host=1.1.1.1', I can see a row like this on Asterisk log: [Jan 13 11:10:54] ERROR[1834]: chan_sip.c:8718 register_verify: Peer '999' is trying to register, but not configured as host=dynamic [Jan 13 11:10:54] NOTICE[1834]: chan_sip.c:15236 handle_request_register: Registration from '999 sip:9...@85.x.y.z ' failed for '174.129.74.46' - Peer is not supposed to register - while if I put 'host=dynamic' I saw (in sip show peers) the row depicted in (1) and no more errors like above. I suspect there is something wrong with network configuration (firewall, NAT). But this behavior is quite odd to me ... Alberto. PS: the network is at customer's site, so I haven't chance to have a clear look over it... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Lister Sent: martedì 12 gennaio 2010 18.51 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question about SIP registration On Tue, 2010-01-12 at 18:16 +0100, Aggio Alberto wrote: Then I have configured an account as following: [999] type=friend username=999 You don't appear to have a secret= line in there with a password option... or did you snip it? Can someone explain me this kind of behaviour? Is it normal? Can I restrict registration of 999 peer only to SIP UA from network 1.1.1.X? There is an ACL option for the SIP peer which you can add, http://www.voip-info.org/wiki/index.php?page=Asterisk+sip +permit-deny-mask (although there were some issues with this in earlier versions of asterisk.. it should work properly in recent versions.) Rob -- _ -- 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] is roundrobin and rrmemory the same meaning?
On Wed, 2010-01-13 at 10:42 +0800, Zhang Shukun wrote: is there some function used to login a agent automaticlly like agentlogin(agentname,agentpassword,phonenumber)? Depends what version you are running. AgentCallBackLogin() is deprecated and you should not use it. But the feature can be reproduced with dialplan logic. http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd% 20AgentCallbackLogin This is a whole world of pain, as is using Agents in some situations. It is better to use SIP channels. (Agents do not seem to work nicely with a bunch of other features.) It is less flexible. It may be better for you to do this using AddQueueMember and RemoveQueueMember on SIP channels, and program a key (or keys) on the handset to add and remove the member from the queue dynamically instead of adding them as static members in queues.conf. Rob -- _ -- 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] Polycom Mute Problem
A little more information please... the PC501 has how many lines defined(the phone has 3 definable, can be 1,2 or 3)? Calls are SIP or DAHDI or Mixed? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael Sent: Tuesday, January 12, 2010 9:56 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Polycom Mute Problem Hey Yall I have an interesting situation. When a person is on the phone (Polycom 501) and another call hits the phone the phone mutes on the users side not the person outside the asterisk system. It stays mute until the call goes to voicemail. Its like the beep we should get from the call waiting has turned into dead silence. Any ideas Thanks Michael D Mosier __ Information from ESET Smart Security, version of virus signature database 4628 (20091122) __ The message was checked by ESET Smart Security. http://www.eset.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipgate DTMF not detected by Asterisk 1.6 [dtmfmode=rfc2833]
On Wed, Jan 13, 2010 at 2:43 AM, listu...@spamomania.co.uk listu...@spamomania.co.uk wrote: Thanks for that. Looking at the RTP packets I can see two types as you point out. The first appears to be the audio: Real-Time Transport Protocol 10.. = Version: RFC 1889 Version (2) Payload type: ITU-T G.711 PCMU (0) And as you say, the DTMF events are clear to see: RFC 2833 RTP Event Event ID: DTMF One 1 (1) ..00 1010 = Volume: 10 So, as these can be seen in the stream, do I need to tell Asterisk to detect these? Does it not do that when I set: dtmfmode=rfc2833 ??? There are some pretty widely recognized RFC2833 compatibility issues in the SIP/RTP world. Which version of Asterisk are you using? Do you know what kind of equipment your carrier is using? If they are using Asterisk you can try to add rfc2833compensate=yes to their peer entry in sip.conf. The SIP debug, however, will tell you if the remote end is configured to use RFC2833 or not. That's why I was telling you to look for telephone-event in the INVITE from your provider. Keep in mind SIP (most likely) runs over UDP between you and your provider, not TCP. I see a 'telephone-event' : a=rtpmap:101 telephone-event/8000 That's all you need to know. They are configured for RFC2833 and they're sending RFC2833. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipgate DTMF not detected by Asterisk 1.6 [dtmfmode=rfc2833]
On Wed, 2010-01-13 at 10:12 -0500, Kristian Kielhofner wrote: On Wed, Jan 13, 2010 at 2:43 AM, listu...@spamomania.co.uk listu...@spamomania.co.uk wrote: Thanks for that. Looking at the RTP packets I can see two types as you point out. The first appears to be the audio: Real-Time Transport Protocol 10.. = Version: RFC 1889 Version (2) Payload type: ITU-T G.711 PCMU (0) And as you say, the DTMF events are clear to see: RFC 2833 RTP Event Event ID: DTMF One 1 (1) ..00 1010 = Volume: 10 So, as these can be seen in the stream, do I need to tell Asterisk to detect these? Does it not do that when I set: dtmfmode=rfc2833 ??? There are some pretty widely recognized RFC2833 compatibility issues in the SIP/RTP world. I had a nasty feeling something like that was coming :-( Which version of Asterisk are you using? Asterisk 1.6.1.11 Do you know what kind of equipment your carrier is using? If they are using Asterisk you can try to add rfc2833compensate=yes to their peer entry in sip.conf. I've not been able to get that out of them, but I don't *think* It's Asterisk based because they say: Unfortunately, our assistance with Asterisk is extremely limited. For configuration problems you will have to rely on other sources. [http://www.sipgate.co.uk/faq/index.php?do=displayArticlearticle=540qw=asterisk] The SIP debug, however, will tell you if the remote end is configured to use RFC2833 or not. That's why I was telling you to look for telephone-event in the INVITE from your provider. Keep in mind SIP (most likely) runs over UDP between you and your provider, not TCP. I see a 'telephone-event' : a=rtpmap:101 telephone-event/8000 That's all you need to know. They are configured for RFC2833 and they're sending RFC2833. I appreciate this is a 'how long is a piece of string question Kristian, but is there likely to be a way I can fix this? -- _ -- 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] Sipgate DTMF not detected by Asterisk 1.6 [dtmfmode=rfc2833]
On Wed, Jan 13, 2010 at 10:39 AM, listu...@spamomania.co.uk listu...@spamomania.co.uk wrote: ..snip.. I've not been able to get that out of them, but I don't *think* It's Asterisk based because they say: Unfortunately, our assistance with Asterisk is extremely limited. For configuration problems you will have to rely on other sources. [http://www.sipgate.co.uk/faq/index.php?do=displayArticlearticle=540qw=asterisk] Just because they don't offer assistance with Asterisk doesn't mean they don't use it themselves. If you send me a packet capture in PCAP format with SIP+RTP between your system and your carrier I can debug this further. I appreciate this is a 'how long is a piece of string question Kristian, but is there likely to be a way I can fix this? You can try the rfc2833compensate option... Other than that I can't know until I see a packet capture. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FW: [mythtv-users] VMWare on the backend. Viable solution?
I found this on the myth-tv list. Can we do the same thing with asterisk? Cheers, Dean -Original Message- From: mythtv-users-boun...@mythtv.org [mailto:mythtv-users-boun...@mythtv.org] On Behalf Of Kenni Lund Sent: Wednesday, January 13, 2010 11:44 AM To: Discussion about mythtv Subject: Re: [mythtv-users] VMWare on the backend. Viable solution? 2010/1/13 Martin Ravell martin.rav...@rave-tech.com.au: General consensust is that in order to use my PVR-350 I'd need to use PCI Passthrough. Support for PCI Passthrough is via a capability known as DV-t (at least on Intel based systems). The boards that I have been looking at (Gigabyte) do not seem to have this enabled. I have sent an inquiry off to Gigabyte to confirm this and will post any response back to this list. I've just changed my motherboard in my old Core 2 based system to a Gigabyte motherboard. The new motherboard is a Gigabyte EQ45M-S2 with Intel Q45 (VT-D capable) chipset. It doesn't say anything about VT-D anywhere, but if you click Ctrl+F1 in the BIOS, some advanced settings will get activated, including VT-D :-D I've tested it yesterday with a PVR-500 card and passthrough worked perfectly! So if your Gigabyte motherboard uses a VT-D capable chipset, there'll be a good chance that you can activate it with Ctrl+F1 in the BIOS. Best Regards Kenni Lund ___ mythtv-users mailing list mythtv-us...@mythtv.org http://mythtv.org/cgi-bin/mailman/listinfo/mythtv-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] Sipgate DTMF not detected by Asterisk 1.6 [dtmfmode=rfc2833]
On Wed, 2010-01-13 at 11:13 -0500, Kristian Kielhofner wrote: On Wed, Jan 13, 2010 at 10:39 AM, listu...@spamomania.co.uk listu...@spamomania.co.uk wrote: ..snip.. I've not been able to get that out of them, but I don't *think* It's Asterisk based because they say: Unfortunately, our assistance with Asterisk is extremely limited. For configuration problems you will have to rely on other sources. [http://www.sipgate.co.uk/faq/index.php?do=displayArticlearticle=540qw=asterisk] Just because they don't offer assistance with Asterisk doesn't mean they don't use it themselves. If you send me a packet capture in PCAP format with SIP+RTP between your system and your carrier I can debug this further. That may contain sensitive data, such as SIP account/password details - so I'll pass on that, but thanks for the offer. -- _ -- 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] Sipgate DTMF not detected by Asterisk 1.6 [dtmfmode=rfc2833]
On Wed, Jan 13, 2010 at 12:07 PM, listu...@spamomania.co.uk listu...@spamomania.co.uk wrote: That may contain sensitive data, such as SIP account/password details - so I'll pass on that, but thanks for the offer. Even if they are using auth it's challenge response and fairly difficult to reverse engineer, not that I have the time for that. I do however, specialize in debugging DTMF. I always make time for interesting cases. I also own a voice service provider so it's unlikely I'm interested in your sipgate credentials :). Good luck. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Mute Problem
Upgrade the phone. I ran into the same issue a year or so ago. There was some setting that was screwed up in the config file and upgrading to the newest version at the time fixed it. It was something like the call waiting tone being 30 seconds of dead air. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael Sent: Tuesday, January 12, 2010 9:56 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Polycom Mute Problem Hey Yall I have an interesting situation. When a person is on the phone (Polycom 501) and another call hits the phone the phone mutes on the users side not the person outside the asterisk system. It stays mute until the call goes to voicemail. Its like the beep we should get from the call waiting has turned into dead silence. Any ideas Thanks Michael D Mosier __ Information from ESET Smart Security, version of virus signature database 4628 (20091122) __ The message was checked by ESET Smart Security. http://www.eset.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Mute Problem
By upgrade the phone I assume you mean upgrade the bios, not purchase a newer phone? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Peder Sent: Wednesday, January 13, 2010 11:25 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Polycom Mute Problem Upgrade the phone. I ran into the same issue a year or so ago. There was some setting that was screwed up in the config file and upgrading to the newest version at the time fixed it. It was something like the call waiting tone being 30 seconds of dead air. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael Sent: Tuesday, January 12, 2010 9:56 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Polycom Mute Problem Hey Yall I have an interesting situation. When a person is on the phone (Polycom 501) and another call hits the phone the phone mutes on the users side not the person outside the asterisk system. It stays mute until the call goes to voicemail. Its like the beep we should get from the call waiting has turned into dead silence. Any ideas Thanks Michael D Mosier __ Information from ESET Smart Security, version of virus signature database 4628 (20091122) __ The message was checked by ESET Smart Security. http://www.eset.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Polycom Mute Problem
Yes. Newer bootrom and sip image and sip.cfg. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Wednesday, January 13, 2010 12:40 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Polycom Mute Problem By upgrade the phone I assume you mean upgrade the bios, not purchase a newer phone? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Peder Sent: Wednesday, January 13, 2010 11:25 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Polycom Mute Problem Upgrade the phone. I ran into the same issue a year or so ago. There was some setting that was screwed up in the config file and upgrading to the newest version at the time fixed it. It was something like the call waiting tone being 30 seconds of dead air. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael Sent: Tuesday, January 12, 2010 9:56 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Polycom Mute Problem Hey Yall I have an interesting situation. When a person is on the phone (Polycom 501) and another call hits the phone the phone mutes on the users side not the person outside the asterisk system. It stays mute until the call goes to voicemail. Its like the beep we should get from the call waiting has turned into dead silence. Any ideas Thanks Michael D Mosier __ Information from ESET Smart Security, version of virus signature database 4628 (20091122) __ The message was checked by ESET Smart Security. http://www.eset.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.28 intermittent one way audio?
Hi All, I made the decision recently to update some old call servers from 1.2.x to the latest 1.4.28. I spent 2 weeks in the lab testing every production requirement for these voice servers. Nothing too special, SIP in, database lookup, SIP out, a couple of special applications, write CDR to billing server, nothing intensive at all. I also performed stress testing for call capacity, database performance and media handling. So after all the testing was finished, I was very confident this was the right thing to do. This was a plain jane asterisk install from stable release on debian etch, SIP only, no zaptel or dahdi. I shut down one of my 1.2 voice servers and turned up the new 1.4.28 server in it's place. Within a couple of hours I started getting calls from a couple of customers complaining about intermittent one-way audio issues, they could her the caller, but the caller could not hear them. This was only happening on inbound calls. I collected call examples for the next few days. I was able to trace almost all the calls back to the new 1.4.28 server. No other changes in the infrastructure other than putting in the 1.4.28 server. And the new server is built on the same hardware as the 1.2 servers are on. I could not find any relevant errors on any associated network element or upstream/downstream voice server, I could not find any errors on the 1.4.28 server either. No errors anywhere, nothing, nada, no retransmits, critical packets, rtp warnings, zilch. I pulled the 1.4.28 server out of call rotation, no more customer complaints. The call volume passing through the server is not a lot, 40 to 80 active calls at any time, and maybe 1 to 2 calls per second max, usually a call every 3 or 4 seconds. The one way audio issue occurred at both high and low call volume and was truly random. I captured 14 calls out of several thousand. I'm sure there was a lot more that that, but these were the only ones reported to me for investigation. So my question is, has anyone else experienced intermittent one way audio specific to Asterisk 1.4 that can be identified and resolved. Or maybe suggest another version of 1.4 that does not have an issue like this at these volumes? Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk / NEC2400 / PRI
Hello List I'm trying to figure out what is wrong between my asterisk and my NEC 2400 pbx We have been trying to link them with a spare PA-24DTG from the NEC, I'm able to call an extension on the Asterisk, however the extension rings, and then immediatly hangs up I traced it back to the debug of the PRI on the Asterisk... I would appreciate if anyone could pin point what is wrong The error code: Cause: Mandatory information element is missing (96), does not tell me what is missing, so any expert outthere who could give me some direction would be extremely helpfull. dadhichannel.conf context=from-internal switchtype = national signalling = pri_net channel = 1-23 context = default group = 63 system.conff span=1,1,0,esf,b8zs # termtype: te bchan=1-23 dchan=24 echocanceller=mg2,1-23 Thanks *Trace:* Enabled debugging on span 1 Protocol Discriminator: Q.931 (8) len=27 Call Ref: len= 2 (reference 99/0x63) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) User information layer 1: u-Law (34) [18 04 e9 80 83 01] Channel ID (len= 6) [ Ext: 1 IntID: Explicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: As indicated in following octets Ext: 1 DS1 Identifier: 0 Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 0 Channel: 1 ] [1e 02 80 81] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Progress Description: Call is not end-to-end ISDN; further call progress information may be available inband. (1) ] [70 05 a1 35 30 30 30] Called Number (len= 7) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '5000' ] -- Making new call for cr 99 -- Processing Q.931 Call Setup -- Processing IE 4 (cs0, Bearer Capability) -- Processing IE 24 (cs0, Channel Identification) -- Processing IE 30 (cs0, Progress Indicator) -- Processing IE 112 (cs0, Called Party Number) q931.c:3551 q931_receive: call 99 on channel 1 enters state 6 (Call Present) q931.c:2816 q931_call_proceeding: call 99 on channel 1 enters state 9 (Incoming Call Proceeding) Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 99/0x63) (Terminator) Message type: CALL PROCEEDING (2) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: As indicated in following octets Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 99/0x63) (Originator) Message type: STATUS (125) [08 03 81 e4 18] Cause (len= 5) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Invalid information element contents (100), class = Protocol Error (e.g. unknown message) (6) ] Cause data 1: 18 (24) [14 01 01] Call State (len= 3) [ Ext: 0 Coding: CCITT (ITU) standard (0) Call state: Call Initiated (1) -- Processing IE 8 (cs0, Cause) -- Processing IE 20 (cs0, Call State) q931.c:2844 q931_alerting: call 99 on channel 1 enters state 7 (Call Received) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 99/0x63) (Terminator) Message type: ALERTING (1) [1e 02 81 88] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 99/0x63) (Originator) Message type: RELEASE (77) [08 03 81 e0 18] Cause (len= 5) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Mandatory information element is missing (96), class = Protocol Error (e.g. unknown message) (6) ] Cause data 1: 18 (24) -- Processing IE 8 (cs0, Cause) q931.c:3801 q931_receive: call 99 on channel 1 enters state 0 (Null) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release Request Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 99/0x63) (Terminator) Message type: RELEASE COMPLETE (90) [08 02 81 e0] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Mandatory information element is missing (96), class = Protocol Error (e.g. unknown message) (6) ] NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the
Re: [asterisk-users] AudioCodes MP-114 - SAS (Stand Alone Survivability) configuration
SOLVED! Correct me anybody if I'm wrong but I think SAS option is for WAN only not for the case if AudioCodes MP and Asterisk are on the same network. I was trying to configure the fail-over mode in scenarios: - Asterisk sever goes down (doesn't happen very often, never happened to me but it could) - hardware failure - power loss etc. (It seems to me Audiocodes caters MS and/or close source hardware; they have many doc files on their web-page but very little or none pertaining to Asterisk configuration with their hardware MediaPack), so here is my solution without going through 500-pages manual. So in the above cases the calls should go through IN or OUT without interuptions, Audiocodes MP can be configured for fail-over mode; my old Liksys 3102 CAN NOT, when power goes down and Linksys is up, the calls will not go through. Here is the relevant configuration for the above scenario: == Protocol Configuration - Proxies/IpGroups/Registration - Proxy Registration - Use Default Proxy: NO Enable Fallback to Routing Table: Enable Prefer Routing Table: NO Enable Registration: Enable Registrar IP Address: 10.0.0.109 (enter IP address of the asterisk server, very important) Gateway Name: (none) Gateway Registration Name: (none) Subscription Mode: Per Endpoint User Name: (none) Password: Default_Passwd Cnonce: Default_Cnonce Authentication Mode: Per Endpoint Routing Tables - Routing General Parameters - Alt Routing Tel to IP Connectivity Method: SIP OPTION ; (in case asterisk and MP-114 are on the same network) Tel to IP Routing - (your numbers and IP's will be different) Src. Trunk Group ID Dest. Phone Prefix Source Phone Prefix Dest. IP Address *369 *10.0.0.157 ***10.0.0.109 ***10.0.0.157 First line is for fax extension, there is no reason to forward it to asterisk, it should go directly to Hylafax (or your fax machine) regardless of asterisk status. Without playing/entering all internal extensions, second line directs all calls to Asterisk sever. Third line is for fail-over mode in case of Asterisk failure, power failure, all calls are directed via to MP-114 so calls go IN and OUT without interruption In addition to these setting user must configure: Routing Tables - IP to Trunk Group Routing Endpoint Settings - Authentication Endpoint Settings - Automatic Dialing Endpoint Number - EndPoint Phone Number Hunt Group - Hunt Group Settings and Optionally: Manipulation Tables - Dest Number IP-Tel === -- Joseph -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] is there some Chinese version of sounds available?
hi ,all when use the VoiceMail , all the directions all english. i want to know is there some Chinese version of sounds available now? or should i record it myself? just like: Here is what you can do with your mailbox using VoiceMailMain. 1 Old Messages 3 Advanced options 1 Send reply 2 Call back 3 Envelope 4 Outgoing call 5 Leave message * Return to main menu 4 Play previous message 5 Repeat current message 6 Play next message 7 Delete current message 8 Forward message to another mailbox 9 Save message in a folder * Help; during msg playback: Rewind # Exit; during msg playback: Fastforward 2 Change folders 3 Advanced options 0 Mailbox options 1 Record your unavailable message 2 Record your busy message 3 Record your name 4 Change your password * Return to the main menu * Help # Exit -- Best regards, Sucan -- _ -- 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] is roundrobin and rrmemory the same meaning?
Thanks for your reply. I have read the Asterisk Realtime Architecture feature of Asterisk. it says that we can save queue and queue_members in the database. and queue_member don't need to login( because not support). and when queue_member changed in database. don't need reload cant asterisk use soon. Has anyone deployed the ARA feature of Asterisk? and How do you think about this feature? 2010/1/13 Robert Lister r...@lentil.org: On Wed, 2010-01-13 at 10:42 +0800, Zhang Shukun wrote: is there some function used to login a agent automaticlly like agentlogin(agentname,agentpassword,phonenumber)? Depends what version you are running. AgentCallBackLogin() is deprecated and you should not use it. But the feature can be reproduced with dialplan logic. http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd% 20AgentCallbackLogin This is a whole world of pain, as is using Agents in some situations. It is better to use SIP channels. (Agents do not seem to work nicely with a bunch of other features.) It is less flexible. It may be better for you to do this using AddQueueMember and RemoveQueueMember on SIP channels, and program a key (or keys) on the handset to add and remove the member from the queue dynamically instead of adding them as static members in queues.conf. Rob -- _ -- 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 -- Best regards, Sucan -- _ -- 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] is there some Chinese version of sounds available?
when use the VoiceMail , all the directions all english. i want to know is there some Chinese version of sounds available now? or should i record it myself? http://www.voip-info.org/wiki/view/Asterisk+sound+files+international Look under Chinese (Mandarin) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ISDN Cause codes for unanswered calls
I am wanting to use the ISDN cause code on an Asterisk 1.6 server to determine the status of a call attempt, where the call might not actually connect. Reason is I am checking for valid telephone numbers from a list of numbers, and I would like to know if the call has answered and cleared which I can by writing the hangupcause variable, but where I get an out of order network message, or number doesn't exist, I want to capture these ISDN cause codes where the call might not have connected and started the dialplan extension. Is there any way to capture cause codes from calls that didn't connect? Thanks Steve Moran Sydney, Australia -- _ -- 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] is there some Chinese version of sounds available?
Thank you! 2010/1/14 Lee, John (Sydney) john@compuware.com: when use the VoiceMail , all the directions all english. i want to know is there some Chinese version of sounds available now? or should i record it myself? http://www.voip-info.org/wiki/view/Asterisk+sound+files+international Look under Chinese (Mandarin) -- _ -- 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 -- Best regards, Sucan -- _ -- 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
