Re: [asterisk-dev] compile asterisk in arm-linux
This type of question should be directed to the asterisk-users list from now on... That being said, a search for asterisk termcap support not found on google yields plenty of results. All suggesting that installing ncurses and ncurses-devel will resolve this problem. On 5/10/07, lizhong zhu [EMAIL PROTECTED] wrote: hello, all asteriskers: i want to compile asterisk under arm-linux. i make a change in Makefile. some errors came out. i have an other asterisk in my system, it works, which means all support should be ready. anyone knows this problem? checking for gcc... /usr/local/arm-linux/bin/arm-linux-gcc checking whether the C compiler (/usr/local/arm-linux/bin/arm-linux-gcc ) works... yes checking whether the C compiler (/usr/local/arm-linux/bin/arm-linux-gcc ) is a cross-compiler... yes checking whether we are using GNU C... yes checking whether /usr/local/arm-linux/bin/arm-linux-gcc accepts -g... yes checking how to run the C preprocessor... /usr/local/arm-linux/bin/arm-linux-gcc -E checking host system type... i686-pc-linux-gnu checking for a BSD compatible install... install checking for ranlib... /usr/local/arm-linux/bin/arm-linux-ranlib checking for ar... /usr/local/arm-linux/bin/arm-linux-ar checking for tgetent in -ltermcap... no checking for tgetent in -ltinfo... no checking for tgetent in -lcurses... no checking for tgetent in -lncurses... no configure: error: termcap support not found make: *** [editline/libedit.a] Error 1 thanks! zhulizhong -- 抢注雅虎免费邮箱3.5G容量，20M附件！ http://cn.mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Asterisk channels limit
Thank you for all. All information was great! Regards On 5/9/07, Kevin P. Fleming [EMAIL PROTECTED] wrote: Jared Smith wrote: As far as I know, there are no hard-coded channel limits in the *open source* version of Asterisk. As I understand it, Digium's Business There wouldn't be any point in having a channel limit in open source Asterisk, since it is by definition open source and users would just remove the limits :-) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev -- -- Paulo Garcia Pika Technologies Inc ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] call transfer to asterik.. asterisk as an end point
Hello All. I am having some trouble with call transfers when asterisk is the 2nd party called and I hope to benefit from your experience. I want to use asterisk for call park/pickup and have configured openser to relay calls made to ruri 700-720 to asterisk running on localhost:5069 Call flow: phone A calls phone B (both phones are polycom) Phone B answers then phone b user presses transfer and dials 700 asterisk plays back 701 as the parking lot location phone B user presses transfer again. at this time phone b is not disconnected from asterisk system phone A is also connected to asterisk and hears 702 as the parking lot location (as if asterisk places the user at priority 1 for that context) From phone C calling 702 will connect phone C to phone A. This was a specific example but this transfer problem is not limited to call park only. It happens any time asterisk is the second party called in call transfer. Thanks in advance for your help. -- Zahid On May 8, 2007, at 1:56 PM, Christian Schlatter wrote: I think I found out why this doesn't work as expected. After phone 1 receives REFER from phone 2, it sends a new INVITE to the asterisk server. This INVITE includes a Replaces: header that tells the receiver (asterisk) to replace an existing SIP dialog with the new one. RFC 3891 The SIP Replaces Header, Section 3 UAS Behavior, defines: the UA attempts to accept the new INVITE, reassign the user interface and other resources of the matched dialog to the new INVITE, and shut down the replaced dialog. But your SIP trace shows that asterisk doesn't shut down the replaced dialog (by sending a BYE), which is the reason why phone 2 does not get disconnected after hitting transfer the second time. Instead of creating a new call park slot (702) when phone 1 sends the Replaces: INVITE to asterisk, asterisk should be intelligent enough to figure out that this INVITE actually replaces the existing SIP dialog with phone 2. And asterisk should not create a new park slot 702 but directly put phone 1 on hold at park slot 701 and send a BYE to phone 2. Although asterisk supports the Replaces: header when used e.g. as a gateway, I have some doubts that the call park/pickup implementation does so too. Especially since it was designed to be used in PBX mode where asterisk acts as B2BUA for all involved call legs. Maybe this should be opened as a new feature/bug request on the asterisk bug tracker. Or maybe there is a asterisk setting that controls this behavior, I'm not really an asterisk expert myself ;-) -- The fact that an opinion has been widely held is no evidence that it is not utterly absurd; indeed, in view of the silliness of the majority of mankind, a widespread belief is more often likely to be foolish than sensible. -Bertrand Russell 8:00? 8:25? 8:40? Find a flick in no time with the Yahoo! Search movie showtime shortcut. http://tools.search.yahoo.com/shortcuts/#news___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re:[asterisk-dev] Response to Subscribe with Expire set to
Date: Thu, 10 May 2007 08:44:08 +0200 From: Olle E Johansson [EMAIL PROTECTED] Subject: Re: [asterisk-dev] Response to Subscribe with Expire set to 0. To: Asterisk Developers Mailing List asterisk-dev@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed Which version are you using? Can you provide me with a SIP debug? Hi Olle, I am using version 1.2.17. The show version CLI gives:Asterisk SVN --r127M built by root @ Ray_dev_VM on a i686 running Linux on 2007-05-10 08:35:40 UTC The following is the sip debug msg I caught: -- SIP read from 10.10.150.164:5060: SUBSCRIBE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.10.150.164;branch=z9hG4bKZgUGdQQPSIVL4XzkREhel0dT8t.C Max-Forwards: 70 From: sip:[EMAIL PROTECTED];tag=zx0blk2XNgw To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 49310 SUBSCRIBE Contact: sip:[EMAIL PROTECTED] Accept: application/simple-message-summary Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO Allow-Events: dialog,message-summary Content-Disposition: session Event: message-summary Expires: 3600 Supported: replaces User-Agent: VoipLan Gateway/07.63 (MAC=00087B030A9F; SER= 02605; HW=12) Content-Type: text/plain Content-Length: 0 --- (18 headers 0 lines) --- May 13 03:38:29 DEBUG[28558]: chan_sip.c:3193 sip_alloc: Allocating new SIP dialog for [EMAIL PROTECTED] - SUBSCRIBE (No RTP) Using latest SUBSCRIBE request as basis request Sending to 10.10.150.164 : 5060 (NAT) Found peer '9002' Looking for 10.10.150.154 in custom (domain 10.10.150.154) Transmitting (NAT) to 10.10.150.164:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.150.164;branch=z9hG4bKZgUGdQQPSIVL4XzkREhel0dT8t.C;received=10.10.150.164 From: sip:[EMAIL PROTECTED];tag=zx0blk2XNgw To: sip:[EMAIL PROTECTED];tag=as6c1cb383 Call-ID: [EMAIL PROTECTED] CSeq: 49310 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 0 Content-Length: 0 --- Destroying call '[EMAIL PROTECTED]' w v -- SIP read from 10.10.150.164:5060: SUBSCRIBE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.10.150.164;branch=z9hG4bKvBdoBF4DKyUrazDA08E5 Max-Forwards: 70 From: sip:[EMAIL PROTECTED];tag=Hps1S2unWoEq To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 9794 SUBSCRIBE Contact: sip:[EMAIL PROTECTED] Accept: application/simple-message-summary Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO Allow-Events: dialog,message-summary Content-Disposition: session Event: message-summary Expires: 3600 Supported: replaces User-Agent: VoipLan Gateway/07.63 (MAC=00087B030A9F; SER= 02605; HW=12) Content-Type: text/plain Content-Length: 0 --- (18 headers 0 lines) --- May 13 03:38:30 DEBUG[28558]: chan_sip.c:3193 sip_alloc: Allocating new SIP dialog for [EMAIL PROTECTED] - SUBSCRIBE (No RTP) Using latest SUBSCRIBE request as basis request Sending to 10.10.150.164 : 5060 (NAT) Found peer '9002' Looking for 10.10.150.154 in custom (domain 10.10.150.154) Transmitting (NAT) to 10.10.150.164:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.150.164;branch=z9hG4bKvBdoBF4DKyUrazDA08E5;received=10.10.150.164 From: sip:[EMAIL PROTECTED];tag=Hps1S2unWoEq To: sip:[EMAIL PROTECTED];tag=as3efaa043 Call-ID: [EMAIL PROTECTED] CSeq: 9794 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 0 Content-Length: 0 --- Destroying call '[EMAIL PROTECTED]' ersion Asterisk SVN--r127M built by root @ Ray_dev_VM on a i686 running Linux on 2007-05-13 08:35:40 UTC *CLI -- SIP read from 10.10.150.164:5060: SUBSCRIBE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.10.150.164;branch=z9hG4bKtCU8wBJeHio.QuanNy60lH6Ynj4UT14 Max-Forwards: 70 From: sip:[EMAIL PROTECTED];tag=0xhXUTUuC To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 47467 SUBSCRIBE Contact: sip:[EMAIL PROTECTED] Accept: application/simple-message-summary Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO Allow-Events: dialog,message-summary Content-Disposition: session Event: message-summary Expires: 3600 Supported: replaces User-Agent: VoipLan Gateway/07.63 (MAC=00087B030A9F; SER= 02605; HW=12) Content-Type: text/plain Content-Length: 0 --- (18 headers 0 lines) --- May 13 03:38:31 DEBUG[28558]: chan_sip.c:3193 sip_alloc: Allocating new SIP dialog for [EMAIL PROTECTED] - SUBSCRIBE (No RTP) Using latest SUBSCRIBE request as basis request Sending to 10.10.150.164 : 5060 (NAT) Found peer '9002' Looking for 10.10.150.154 in custom (domain 10.10.150.154) Transmitting (NAT) to 10.10.150.164:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.150.164;branch=z9hG4bKtCU8wBJeHio.QuanNy60lH6Ynj4UT14;received=10.10.150.164 From: sip:[EMAIL PROTECTED];tag=0xhXUTUuC To: sip:[EMAIL PROTECTED];tag=as0a0daba8 Call-ID: [EMAIL PROTECTED]
[asterisk-dev] chan_cellphone - Mantis issue 8919
Hey all, This was discussed a little in the past, but I'd like to bring it up one more time, and get an official consensus from the community. I'm planning on committing the patch in Mantis issue 8919 (chan_cellphone) soon, and before I do, I wanted to get thoughts from all of you on what we should actually name it. Some of you (myself included) have voiced concerns about the name of the channel driver - it's outgrown it, since it no longer supports *just* cellphones. It also supports bluetooth headsets, and in the future could support even more (bluetooth fax profile, anybody? who knows what the future may hold). chan_bluetooth seems to be the most obvious name, but that's already used by another similar channel driver - that would only cause confusion...and confusion is very bad. I'd like to propose the name chan_mobile (Matthew Rubenstein gets credit for suggesting it as the new name (but a google search says I called it 6 months prior - before chan_cellphone even existed :p)) If you have thoughts/feelings for/against the name or for/against the name change, please feel free to voice them here, or to me privately. -- Jason Parker Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] chan_cellphone - Mantis issue 8919
10 maj 2007 kl. 20.18 skrev Jason Parker: Hey all, This was discussed a little in the past, but I'd like to bring it up one more time, and get an official consensus from the community. I'm planning on committing the patch in Mantis issue 8919 (chan_cellphone) soon, and before I do, I wanted to get thoughts from all of you on what we should actually name it. Some of you (myself included) have voiced concerns about the name of the channel driver - it's outgrown it, since it no longer supports *just* cellphones. It also supports bluetooth headsets, and in the future could support even more (bluetooth fax profile, anybody? who knows what the future may hold). chan_bluetooth seems to be the most obvious name, but that's already used by another similar channel driver - that would only cause confusion...and confusion is very bad. I'd like to propose the name chan_mobile (Matthew Rubenstein gets credit for suggesting it as the new name (but a google search says I called it 6 months prior - before chan_cellphone even existed :p)) If you have thoughts/feelings for/against the name or for/against the name change, please feel free to voice them here, or to me privately. chan_pan :-) PAN/nokia PAN/peter PAN = Personal Area Network /O ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] chan_cellphone - Mantis issue 8919
On Thursday 10 May 2007 2:26 pm, Olle E Johansson wrote: chan_pan :-) PAN/nokia PAN/peter PAN = Personal Area Network Eww... this isn't using the PAN profile at all, so I don't think that'd be right... chan_HFP or HSP would be my guess if you wanted to go that route... but yeah chan_bluetooth should really be what this should be called, since it'll handle damn near everything. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] chan_cellphone - Mantis issue 8919
Andrew Kohlsmith wrote: On Thursday 10 May 2007 2:26 pm, Olle E Johansson wrote: chan_pan :-) PAN/nokia PAN/peter PAN = Personal Area Network Eww... this isn't using the PAN profile at all, so I don't think that'd be right... chan_HFP or HSP would be my guess if you wanted to go that route... but yeah chan_bluetooth should really be what this should be called, since it'll handle damn near everything. Name it something totally off the wall like chan_cthulhu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] chan_cellphone - Mantis issue 8919
On Thursday 10 May 2007 14:35, Andrew Kohlsmith wrote: On Thursday 10 May 2007 2:26 pm, Olle E Johansson wrote: chan_pan :-) PAN/nokia PAN/peter PAN = Personal Area Network Eww... this isn't using the PAN profile at all, so I don't think that'd be right... chan_HFP or HSP would be my guess if you wanted to go that route... but yeah chan_bluetooth should really be what this should be called, since it'll handle damn near everything. We could go plural... chan_blueteeth. -- Tilghman ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] chan_cellphone - Mantis issue 8919
Tilghman Lesher wrote: We could go plural... chan_blueteeth. +1 !!! -- Russell Bryant Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] RTP Bridging optimization
I wonder if somebody considered to optimize RTP bridging using spllice and tee syscalls? Vadim ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] RTP bridiging optimization
Hello I wonder if somebody cosnidered to optimize rtp bridging using Linux splice and tee syscalls? Thanks Vadim ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] RTP bridiging optimization
Hello I wonder if somebody considered to optimize rtp bridging using Linux splice and tee syscalls? Thanks Vadim ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] RTP bridiging optimization
I wonder if you considered sending the same message to the same mailing list 3 times in a row? (Yes, I know I am not helpful.) On 5/10/07, Vadim Lebedev [EMAIL PROTECTED] wrote: Hello I wonder if somebody considered to optimize rtp bridging using Linux splice and tee syscalls? Thanks Vadim ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
