Re: [asterisk-users] Call Transfer Fails - Not a Valid Extension
The file /var/log/asterisk/full will contain helpful log messages that show how Asterisk is internally handling the call. It may be necessary to increase the verbosity of the log to get more details however. From the linux command line, you can follow these steps to get a copy of the relevant messages: # asterisk -rx core set verbose 5 # cat /var/log/asterisk/full mylogfile (perform a transfer that fails with the message now, then press CTRL-C to cancel the above command) The mylogfile will have the log entries necessary to understand what happened, although it may also require an understanding of the FreePBX dialplan to interpret it. If you can post your log file (recommend using a pastebin rather than emailing the whole thing) it should be fairly easy to spot the problem and advise you how to fix it. On Sun, Sep 7, 2014 at 10:55 PM, Phil Ledon ple...@lodgetech.com wrote: We have a plain vanilla installation of AsteriskNOW using Digium D40/50 phones. All transfers are failing from any source to any extension with the message “that is not a valid extension”. Does anyone have any ideas about where to begin looking for the source of that error? *Phil Ledon* -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out at: http://digium.com · http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Transfer Fails - Not a Valid Extension
We have a plain vanilla installation of AsteriskNOW using Digium D40/50 phones. All transfers are failing from any source to any extension with the message that is not a valid extension. Does anyone have any ideas about where to begin looking for the source of that error? Phil Ledon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call transfer problem.
You have to use attendant transfer, not blind. - A calls B - B answers on line 1 (button 1) - B has to use line 2 (push button 2) to call C, C sees call coming from B, the same does asterisk - while having line 2 active, he pushes button transfer followed by button line 1 - A speaks with C On Mon, Feb 24, 2014 at 7:45 PM, Mike Diehl mdiehlena...@gmail.com wrote: I'm sorry, I should have mentioned that he's doing a phone-based transfer, not an asterisk-based transfer. Mike. On Mon, Feb 24, 2014 at 1:30 PM, Don Kelly d...@donkelly.biz wrote: Does he complete the call as a supervised transfer--waits for the called party to answer before completing the transfer? --Don -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl Sent: Monday, February 24, 2014 12:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Call transfer problem. Hi all, I have a user who is having trouble transferring calls, using a Grandstream GXP2xxx. Here's the use case that I've seen: I call the user from phone A and he answers on phone B. Then, he hits the transfer button on his phone and dials an extension that is reachable by him, but not by me, based on administrative policy. However, the Asterisk logs indicate that the new call is being initiated by phone A, not phone B! Thus the call transfer fails. I have other users, with other phones, that are able to transfer just fine. What could be different with this particular user? Any ideas? Mike. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call transfer problem.
Hi all, I have a user who is having trouble transferring calls, using a Grandstream GXP2xxx. Here's the use case that I've seen: I call the user from phone A and he answers on phone B. Then, he hits the transfer button on his phone and dials an extension that is reachable by him, but not by me, based on administrative policy. However, the Asterisk logs indicate that the new call is being initiated by phone A, not phone B! Thus the call transfer fails. I have other users, with other phones, that are able to transfer just fine. What could be different with this particular user? Any ideas? Mike. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call transfer problem.
I'm sorry, I should have mentioned that he's doing a phone-based transfer, not an asterisk-based transfer. Mike. On Mon, Feb 24, 2014 at 1:30 PM, Don Kelly d...@donkelly.biz wrote: Does he complete the call as a supervised transfer--waits for the called party to answer before completing the transfer? --Don -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl Sent: Monday, February 24, 2014 12:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Call transfer problem. Hi all, I have a user who is having trouble transferring calls, using a Grandstream GXP2xxx. Here's the use case that I've seen: I call the user from phone A and he answers on phone B. Then, he hits the transfer button on his phone and dials an extension that is reachable by him, but not by me, based on administrative policy. However, the Asterisk logs indicate that the new call is being initiated by phone A, not phone B! Thus the call transfer fails. I have other users, with other phones, that are able to transfer just fine. What could be different with this particular user? Any ideas? Mike. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call transfer problem.
Does he complete the call as a supervised transfer--waits for the called party to answer before completing the transfer? --Don -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl Sent: Monday, February 24, 2014 12:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Call transfer problem. Hi all, I have a user who is having trouble transferring calls, using a Grandstream GXP2xxx. Here's the use case that I've seen: I call the user from phone A and he answers on phone B. Then, he hits the transfer button on his phone and dials an extension that is reachable by him, but not by me, based on administrative policy. However, the Asterisk logs indicate that the new call is being initiated by phone A, not phone B! Thus the call transfer fails. I have other users, with other phones, that are able to transfer just fine. What could be different with this particular user? Any ideas? Mike. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Transfer question
Hi, is possible that two sip extensions: user-1 and user-2 are connected and I want to transfer the call from user-1 to a third user user-3. I know it is possible through feature keys mapping in features.conf, but I want to do this through AMI or Asterisk CLI Commands? Please suggest if possible? Thank you! Muhammad Faheem -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Transfer question
Hi faheem, You can do this: ACTION: Redirect Channel: Channel ID Context: Context Exten: Exten Priority: Priority Regards, Qasim On Thu, May 16, 2013 at 3:13 PM, Muhammad Faheem faheem2...@gmail.comwrote: Hi, is possible that two sip extensions: user-1 and user-2 are connected and I want to transfer the call from user-1 to a third user user-3. I know it is possible through feature keys mapping in features.conf, but I want to do this through AMI or Asterisk CLI Commands? Please suggest if possible? Thank you! Muhammad Faheem -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Transfer not working
On 9/4/12 3:04 am, Takehiro Matsushima wrote: // I don't know what's difference t and T. T allows the caller to transfer. t allows the called user to transfer. You very rarely want Tt - since I doubt you want an incoming caller to be able to transfer their call all over the place. You usually want t on incoming calls and T on outgoing calls. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Transfer not working
Thank you so much. OK, I understood that to transfer the call t is usually used, is it right? And I should write so in my last mail. t and T are described with same sentences in official wiki... Regards, Takehiro Matsushima 2012/4/9 Chris Bagnall aster...@lists.minotaur.cc: On 9/4/12 3:04 am, Takehiro Matsushima wrote: // I don't know what's difference t and T. T allows the caller to transfer. t allows the called user to transfer. You very rarely want Tt - since I doubt you want an incoming caller to be able to transfer their call all over the place. You usually want t on incoming calls and T on outgoing calls. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Transfer not working
Thanks everyone. I was using the Tt flag but in the wrong place in the dial application. Cheers On Mon, Apr 9, 2012 at 4:54 PM, Takehiro Matsushima takehiro.dream...@gmail.com wrote: Thank you so much. OK, I understood that to transfer the call t is usually used, is it right? And I should write so in my last mail. t and T are described with same sentences in official wiki... Regards, Takehiro Matsushima 2012/4/9 Chris Bagnall aster...@lists.minotaur.cc: On 9/4/12 3:04 am, Takehiro Matsushima wrote: // I don't know what's difference t and T. T allows the caller to transfer. t allows the called user to transfer. You very rarely want Tt - since I doubt you want an incoming caller to be able to transfer their call all over the place. You usually want t on incoming calls and T on outgoing calls. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Transfer not working
Hi. Maybe you forgotten specify to allow the transferring a call. Try with tT options in Dial() in extensions.conf. // I don't know what's difference t and T. -- Takehiro Matsushima takehiro.dream...@gmail.com 2012/4/7 Rizwan Hisham rizwanhas...@gmail.com: Hi All, I am using asterisk 1.8.11 on centos 5. I have realtime sip peers with dtmf setting rfc2833 and inband. I have also enabled blind and attended transfer features in features.conf but still call transfers dont work. I have setup transfer feature in past but i dont think i am missing anything this time. I just dont have any clue why its not working. I have tried using ATAs and softphones but cant make it to work. Can anyone help? Am I missing anything? features show output: === Builtin Feature Default Current --- --- --- Pickup *8 *8 Blind Transfer # #1 Attended Transfer *2 One Touch Monitor Disconnect Call * * Park Call One Touch MixMonitor == -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] call transfer back to a sourcing switch
If call comes into PBX-A and based on the DNIS it comes into my box PBX-B my box then says ring phone C. Person answers. They want to transfer the call to a phone going back out PBX-A. All this is fine of course. my question is when phone C transfers the call is there a way PBX-B can drop out of the mix. Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] call transfer
call transfer call transfer from reception to other extensions. Question: Details of Extensions Reception - 2000 Sales - 2001 Accounts - 2002 any call comes it should be received by extenion 2000, n if person wants to talk to Sales, receptionist should put the caller on hold than connect to Sales i.e exten 2001, while on hold the caller should hear music on hold,now sale exten can take his call n talk to it.same with Accounts ext 2002. vi /etc/asterisk/sip.conf [general] port = 5060 bindaddr = 0.0.0.0 context = others [2000] type=friend context=reception secret=1234 host=dynamic [2001] type=friend context=sales secret=1234 host=dynamic [2002] type=friend context=accounts secret=1234 host=dynamic ~ ## vi /etc/asterisk/extension.conf [from-zaptel] exten = s,1,wait(2) exten = s,n,Dial(SIP/2000,20) what to next Your Mail works best with the New Yahoo Optimized IE8. Get it NOW! http://downloads.yahoo.com/in/internetexplorer/-- _ -- 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] call transfer
On Tue, 2010-02-16 at 17:25 +0530, cool dude wrote: call transfer call transfer from reception to other extensions. Question: Details of Extensions Reception - 2000 Sales - 2001 Accounts - 2002 any call comes it should be received by extenion 2000, n if person wants to talk to Sales, receptionist should put the caller on hold than connect to Sales i.e exten 2001, while on hold the caller should hear music on hold,now sale exten can take his call n talk to it.same with Accounts ext 2002. vi /etc/asterisk/sip.conf [general] port = 5060 bindaddr = 0.0.0.0 context = others [2000] type=friend context=reception secret=1234 host=dynamic [2001] type=friend context=sales secret=1234 host=dynamic [2002] type=friend context=accounts secret=1234 host=dynamic ~ ## vi /etc/asterisk/extension.conf [from-zaptel] exten = s,1,wait(2) exten = s,n,Dial(SIP/2000,20) what to next __ What have you tried? Which links/mans have you read to set up music on hold? Are any of them wrong at all? -- _ -- 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] call transfer
Tuesday, February 16, 2010, 12:55:12 PM, cool wrote: call comes it should be received by extenion 2000, n if person wants to talk to Sales, receptionist should put the caller on hold than connect to Sales i.e exten 2001, while on hold the caller should hear music on hold,now sale exten can take his call n talk to it.same with Accounts ext 2002. ... what to next To have call transfer in your asterisk setup, YOU need to read some documentation. Start here: http://www.voip-info.org -- Best regards, Gergomailto:csi...@gmail.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] Call Transfer Problem
Hello, I am having a problem with getting call transfer to work. This is what is happening:- 1) External call comes in on SIP from a DDI provider 2) The call is answered by extension 204 3) Then extension 204 presses the Xfer button and the call is placed on hold 4) Extension 204 calls extension 201 and speaks to them. 5) Extension 204 presses the xfer button again to complete the transfer. The result is that the caller is cut off and the SIP Debug in asterisk shows the following:- SIP/2.0 481 Call leg/transaction does not exist Below is a clip from the debug list. I would greatly appreciate any help as the client is getting annoyed. Regards Dan -- Packet2Packet bridging SIP/winsor_204-12cb4160 and SIP/winsor_201-12ca50b0 sip1*CLI --- SIP read from 94.193.81.135:49160 --- ACK sip:2...@83.222.226.126 SIP/2.0 Via: SIP/2.0/UDP 94.193.81.135:49160;branch=z9hG4bK-9ba5b149 From: Rachael sip:winsor_...@sip1.keshercommunications.com;tag=127e2c656448055eo0 To: Robert sip:2...@sip1.keshercommunications.com;tag=as1db0f5fd Call-ID: 5060f231-68791...@94.193.81.135 CSeq: 102 ACK Max-Forwards: 70 Proxy-Authorization: Digest username=winsor_204,realm=asterisk,nonce=24eede11,uri=sip:2...@83. 222.226.126,algorithm=MD5,response=a3b443415fd656ce42253002548a823a Contact: Rachael sip:winsor_...@94.193.81.135:49160 User-Agent: Sipura/SPA921-4.1.10(b) Content-Length: 0 - --- (11 headers 0 lines) --- sip1*CLI --- SIP read from 94.193.81.135:49160 --- REFER sip:901617720...@83.222.226.126 SIP/2.0 Via: SIP/2.0/UDP 94.193.81.135:49160;branch=z9hG4bK-5479aeea From: sip:winsor_...@94.193.81.135:49160;tag=f2c2287b333442fi0 To: 01617720007 sip:901617720...@83.222.226.126;tag=as2eb45d54 Referred-By: Rachael sip:winsor_...@sip1.keshercommunications.com Call-ID: 15dcfde333cdaf86302cb6490b04d...@83.222.226.126 CSeq: 102 REFER Max-Forwards: 70 Contact: Rachael sip:winsor_...@94.193.81.135:49160 efer-To: sip:2...@83.222.226.126?replaces=5060f231%2d68791a02%4010%2e0%2e0%2e204% 3Bfrom-tag%3D127e2c656448055eo0%3Bto-tag%3Das1db0f5fd User-Agent: Sipura/SPA921-4.1.10(b) Content-Length: 0 - --- (12 headers 0 lines) --- Call 15dcfde333cdaf86302cb6490b04d...@83.222.226.126 got a SIP call transfer from caller: (REFER)! SIP transfer to extension 2...@winsor_phones by winsor_...@sip1.keshercommunications.com --- Transmitting (NAT) to 94.193.81.135:49160 --- SIP/2.0 202 Accepted Via: SIP/2.0/UDP 94.193.81.135:49160;branch=z9hG4bK-5479aeea;received=94.193.81.135 From: sip:winsor_...@94.193.81.135:49160;tag=f2c2287b333442fi0 To: 01617720007 sip:901617720...@83.222.226.126;tag=as2eb45d54 Call-ID: 15dcfde333cdaf86302cb6490b04d...@83.222.226.126 CSeq: 102 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: sip:901617720...@83.222.226.126 Content-Length: 0 set_destination: Parsing sip:winsor_...@94.193.81.135:49160 for address/port to send to set_destination: set destination to 94.193.81.135, port 49160 Reliably Transmitting (NAT) to 94.193.81.135:49160: NOTIFY sip:winsor_...@94.193.81.135:49160 SIP/2.0 Via: SIP/2.0/UDP 83.222.226.126:5060;branch=z9hG4bK2e10dade;rport From: 01617720007 sip:901617720...@83.222.226.126;tag=as2eb45d54 To: sip:winsor_...@94.193.81.135:49160;tag=f2c2287b333442fi0 Contact: sip:901617720...@83.222.226.126 Call-ID: 15dcfde333cdaf86302cb6490b04d...@83.222.226.126 CSeq: 103 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: 01617720007 sip:901617720...@83.222.226.126;privacy=off;screen=no Event: refer;id=102 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 49 SIP/2.0 481 Call leg/transaction does not exist ___ -- 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] call transfer using DTMF
Is there a way to transfer a call, while in the middle of the call, using DTMF? ___ -- 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] call transfer using DTMF
On 15/7/09 3:07 PM, Michael wrote: Is there a way to transfer a call, while in the middle of the call, using DTMF? Yep, just pass the t or T options to the dial command and set it up in /etc/asterisk/features.conf -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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] call transfer using DTMF
Yes, In the features.conf under featuremap you need the blindtransfer un-commented blindxfer = ## Then in your extensions.conf you need to have at least a capital T exten = example,1,Dial(ZAP/4/12345,,T) Then during the call you can press ## and asterisk will say transfer. Then dial in the extension you want to transfer too. Thank you, Brad Finberg - Original Message - From: Michael as...@nettrust.co.nz To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: Date: Tuesday, July 14 2009 11:07 PM Subject: [asterisk-users] call transfer using DTMF Is there a way to transfer a call, while in the middle of the call, using DTMF? ___ -- 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] call transfer in CDR
On Thu, Jan 15, 2009 at 4:09 AM, Rilawich Ango maillist...@gmail.com wrote: Hi, I wonder how I can relate the CDR records for the case of call transfer. I can't find their relationship in CDR. Any can advice? ango You may want to read this thread. http://lists.digium.com/pipermail/asterisk-users/2008-January/204856.html Regards, Greyman. ___ -- 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] call transfer in CDR
Hi, I wonder how I can relate the CDR records for the case of call transfer. I can't find their relationship in CDR. Any can advice? ango ___ -- 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] Call transfer using agi
You could simply have it Dial() to wherever it needs to go at the end of the script. 2009/1/6 Rajkumar S rajkum...@gmail.com Hi, I have a typical call center with queues and agents added via AddQueueMember. One of my requirement is to implement a forgot password function. If a caller does not remember the password, he calls up an unauthenticated line and the agent manually authenticates him. Then the caller should have a provision to reset his password. The requirement is that the agent should not know the new password of caller. I have an agi to change password and can transfer call to agi, but I do not know how to transfer the call back to agent from agi. So basically how can an agi transfer a call to an extension? Thanks and regards, raj ___ -- 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 -- 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
[asterisk-users] Call transfer using agi
Hi, I have a typical call center with queues and agents added via AddQueueMember. One of my requirement is to implement a forgot password function. If a caller does not remember the password, he calls up an unauthenticated line and the agent manually authenticates him. Then the caller should have a provision to reset his password. The requirement is that the agent should not know the new password of caller. I have an agi to change password and can transfer call to agi, but I do not know how to transfer the call back to agent from agi. So basically how can an agi transfer a call to an extension? Thanks and regards, raj ___ -- 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] Call transfer over IAX trunk
Ciao Noah, What flags do you have in your Dial() statement? If you want both parties to be able to transfer with the features.conf transfer, you need to have 'Tt' in your dial statement, like this: Dial(IAX2/user:[EMAIL PROTECTED]/exten,20,Tt) Bingo. That was the problem. Thanks a lot, -- Dr. Andrea Spadaccini Multimedia Technologies Institute - MTI S.r.l. Web: www.x-voice.it - Tel: +39 (0) 95 7224945 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call transfer over IAX trunk
Hi Andrea - I have two asterisk servers, an IAX trunk between and some SIP users registered to each server. The scenario is this: user A, registered to PBX 1, calls user B, registered to PBX 2. Then A wants to transfer the call using the features.conf method (in my case, **), but is unable to do this. What flags do you have in your Dial() statement? If you want both parties to be able to transfer with the features.conf transfer, you need to have 'Tt' in your dial statement, like this: Dial(IAX2/user:[EMAIL PROTECTED]/exten,20,Tt) - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call transfer over IAX trunk
Hello everybody, I have two asterisk servers, an IAX trunk between and some SIP users registered to each server. The scenario is this: user A, registered to PBX 1, calls user B, registered to PBX 2. Then A wants to transfer the call using the features.conf method (in my case, **), but is unable to do this. If B wanted to transfer the call, it would work. If A wanted to transfer the call using the Transfer button in the phone, it would work. But since it is a fairly large installation, I'd like A to be able to transfer using **, as this is the method that has been taught to all users. Any hints? Thanks in advance, -- Dr. Andrea Spadaccini Multimedia Technologies Institute - MTI S.r.l. Web: www.x-voice.it - Tel: +39 (0) 95 7224945 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call transfer over IAX trunk
Hello everybody, I have two asterisk servers, an IAX trunk between and some SIP users registered to each server. The scenario is this: user A, registered to PBX 1, calls user B, registered to PBX 2. Then A wants to transfer the call using the features.conf method (in my case, **), but is unable to do this. If B wanted to transfer the call, it would work. If A wanted to transfer the call using the Transfer button in the phone, it would work. But since it is a fairly large installation, I'd like A to be able to transfer using **, as this is the method that has been taught to all users. Any hints? Thanks in advance, -- Dr. Andrea Spadaccini Multimedia Technologies Institute - MTI S.r.l. Web: www.x-voice.it - Tel: +39 (0) 95 7224945 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Transfer
Hello all I'ts my first message here although I follow the list for about a month now. I'd like to ask a question because googling was not so helpful. Here it is: Is there any way to transfer the Incoming CallerID (the one who called my office) when I transfer the call to an internal extension? The reson I'm asking: I work with an ERP system not made by me. I wrote an external application and catch the called ID when the caller rings my internal number (Direct call)e.g my telephone number is (2810)123456 and ext 54. If I'm called directly the I have the CalledID which is passed to the app and I raise an event to the ERP system to show me the details of the caller (name, etc). If my secretary or anyone else picks up the call when the line is transferred in my ext then I have the internal caller ID. Can I have somehow the External callerID? Thank you very much in advance... Theodore Greece ___ -- 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] Call Transfer
On Sat, Jun 7, 2008 at 8:24 AM, Theodore Patsiouras [EMAIL PROTECTED] wrote: If my secretary or anyone else picks up the call when the line is transferred in my ext then I have the internal caller ID. Can I have somehow the External callerID? Look at the channel variables that contain the callerid information. You can assign the incoming callerid to the one that makes the call to your local extension to do what you wish. ___ -- 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] call transfer issue
Hi, I use asterisk 1.2.23 I have the following issue with transfer: I call from from sipA to sipB when sipB press transfer (not blanktransfer) sipA hear the music until sipB put down the phone, in this time sipC is ringing but sipA don't hear anything can you tell me where to lookup the problem of stop music ? BR, Adi ___ -- 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] call transfer detection in dial plan
On 9/13/07, Rilawich Ango [EMAIL PROTECTED] wrote: Hi all, In default, we can use # to transfer the call. I want to know how I can know the user presse # to transfer the call in dial plan. ango Set TRANSFER_CONTEXT or GOTO_ON_BLINDXFER variable (depending on * version) before Dial(). I just wrote more explaining mail to list. Regards, Atis -- Atis Lezdins, IT Responsible of BEST Riga, [EMAIL PROTECTED] ICQ: 142239285 Skype: atis.lezdins Cell Phone: +371 28806004 [Tele2, Latvia] Work phone: +1 800 7502835 [Toll free, USA] ?BEST? - www.BEST.eu.org ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] call transfer detection in dial plan
Hi all, In default, we can use # to transfer the call. I want to know how I can know the user presse # to transfer the call in dial plan. ango ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] call transfer not working
check to see if you have dtmf=rfc2833 and canreinvite=no in sip.conf general settings. On 7/4/07, satish patel [EMAIL PROTECTED] wrote: Dear all I have install asterisk 1.2.x and it is working fine my setup is like [*]---[Mediant2k][Avaya] Now i want to transfer call in internal extension i have read more document on www.voip-info.com but it is now so much clear so if u have any sample configuration file and doucment plz suggest me i have configure feature.conf and extention.conf for this task regards -- Get the Yahoo! toolbar and be alerted to new email http://us.rd.yahoo.com/evt=48225/*http://new.toolbar.yahoo.com/toolbar/features/mail/index.phpwherever you're surfing. ___ --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 Rizwan Hisham Software Engineer AXVOICE Inc. www.axvoice.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] call transfer not working
Dear all I have install asterisk 1.2.x and it is working fine my setup is like [*]---[Mediant2k][Avaya] Now i want to transfer call in internal extension i have read more document on www.voip-info.com but it is now so much clear so if u have any sample configuration file and doucment plz suggest me i have configure feature.conf and extention.conf for this task regards - Get the Yahoo! toolbar and be alerted to new email wherever you're surfing. ___ --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] Call transfer in asterisk
dear all I am new in asterisk and i have now setup asterik for 40 phone now i want to configure call transfer between phone so how it is possible and what configuration part in asterisk will perfomed for this task give me suggestion for my solution Regards Satish Patel - Yahoo! oneSearch: Finally, mobile search that gives answers, not web links. ___ --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] Call transfer in asterisk
On Monday 02 July 2007 01:45:44 pm satish patel wrote: dear all I am new in asterisk and i have now setup asterik for 40 phone now i want to configure call transfer between phone so how it is possible and what configuration part in asterisk will perfomed for this task give me suggestion for my solution Regards Satish Patel - Yahoo! oneSearch: Finally, mobile search that gives answers, not web links. http://www.voip-info.org/wiki/index.php?page=PBX+CallTransfer -- Dominik Zalewski | System Administrator OpenCraft t- +2 02 336 0003 w- http://www.open-craft.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] Call transfer in asterisk
satish patel wrote: dear all I am new in asterisk and i have now setup asterik for 40 phone now i want to configure call transfer between phone so how it is possible and what configuration part in asterisk will perfomed for this task give me suggestion for my solution Regards Satish Patel And this: http://www.voip-info.org/wiki-Asterisk+config+features.conf Warm Regards, Lee ___ --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] Call transfer in asterisk
Dear all i have read that document but dont understand about function i have include featuremap in extension.conf [mysip] include = featuremap and reload extention.conf i got this error *CLI extensions reload Jul 2 19:23:04 WARNING[16320]: pbx.c:6444 ast_context_verify_includes: Context 'mysip' tries includes nonexistent context 'featuremap' *CLI also i have chenged in feature.conf [featuremap] blindxfer = #1 ; Blind transfer disconnect = *0; Disconnect automon = *1 ; One Touch Record atxfer = *2; Attended transfer why my inculde function not working properly Lee Jenkins [EMAIL PROTECTED] wrote: satish patel wrote: dear all I am new in asterisk and i have now setup asterik for 40 phone now i want to configure call transfer between phone so how it is possible and what configuration part in asterisk will perfomed for this task give me suggestion for my solution Regards Satish Patel And this: http://www.voip-info.org/wiki-Asterisk+config+features.conf Warm Regards, Lee ___ --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 - Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. ___ --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] Call transfer feature
satish patel wrote: Dear ALL I want to transfer call from one phone 2 another phone so this is asterisk feature or SIP Phone feature or endpoint feature how can i transfer phone call from to another phone Rgd Satish patel Check out this page: http://www.voip-info.org/wiki-Asterisk+config+features.conf -- Warm Regards, Lee ___ --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] Call transfer feature
Dear ALL I want to transfer call from one phone 2 another phone so this is asterisk feature or SIP Phone feature or endpoint feature how can i transfer phone call from to another phone Rgd Satish patel - Looking for a deal? Find great prices on flights and hotels with Yahoo! FareChase.___ --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] call transfer problem
Dear ALL I have asterisk with sip and it is integrated with avaya through mediant [*]-[mediant 2000]-E1--[Avaya] Now i want to call transfer feature in asterisk means transfer call from one phone 2 another phone how could it possible with asterisk Regrads Satish - Finding fabulous fares is fun. Let Yahoo! FareChase search your favorite travel sites to find flight and hotel bargains.___ --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] Call transfer while dialing
Hi, I want to transfer the call to a conferencing room while dialing. I tried to do that using manager API(Redirect), but it did't work. Regards, Jason. Don't pick lemons. See all the new 2007 cars at Yahoo! Autos. http://autos.yahoo.com/new_cars.html ___ --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] 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-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CALL TRANSFER
Hi Guys, I'm implementing my Asterisk step by step, so far the communications between softphones, hardphones with Gateways, voice mail, are working fine. Rightnow I would like to enable Call tranfer (like Traditional PBX) in Blind Transfer and AttendXFER, I'm reading features.conf in accordance to voip-info.org but the transfer doesn't work! Please if you can provide me some examples will be very appreciate. Rgds. -- Omar E.P.T - Certified Networking Professionals make better Connections! http://omarept.blogspot.com/ Usysnet Corp Open Source Solutions www.usysnet.com.pe ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] CALL TRANSFER
Your dial string must have either the t or T option set. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of omar parihuana Sent: Friday, December 01, 2006 9:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] CALL TRANSFER Hi Guys, I'm implementing my Asterisk step by step, so far the communications between softphones, hardphones with Gateways, voice mail, are working fine. Rightnow I would like to enable Call tranfer (like Traditional PBX) in Blind Transfer and AttendXFER, I'm reading features.conf in accordance to voip-info.org but the transfer doesn't work! Please if you can provide me some examples will be very appreciate. Rgds. -- Omar E.P.T - Certified Networking Professionals make better Connections! http://omarept.blogspot.com/ Usysnet Corp Open Source Solutions www.usysnet.com.pe ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CALL TRANSFER
Thanks!!! I forget Tt option! (too basis!!) On 12/1/06, Damon Estep [EMAIL PROTECTED] wrote: Your dial string must have either the t or T option set. -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *omar parihuana *Sent:* Friday, December 01, 2006 9:10 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] CALL TRANSFER Hi Guys, I'm implementing my Asterisk step by step, so far the communications between softphones, hardphones with Gateways, voice mail, are working fine. Rightnow I would like to enable Call tranfer (like Traditional PBX) in Blind Transfer and AttendXFER, I'm reading features.conf in accordance to voip-info.org but the transfer doesn't work! Please if you can provide me some examples will be very appreciate. Rgds. -- Omar E.P.T - Certified Networking Professionals make better Connections! http://omarept.blogspot.com/ Usysnet Corp Open Source Solutions www.usysnet.com.pe ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Omar E.P.T - Certified Networking Professionals make better Connections! http://omarept.blogspot.com/ Usysnet Corp Open Source Solutions www.usysnet.com.pe ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] CALL TRANSFER
Be careful, if you set both T and t you might be allowing the wrong party to transfer the call! In MOST cases you would want T or t, not T and t, although there are some cases where you might want both. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of omar parihuana Sent: Friday, December 01, 2006 5:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] CALL TRANSFER Thanks!!! I forget Tt option! (too basis!!) On 12/1/06, Damon Estep [EMAIL PROTECTED] wrote: Your dial string must have either the t or T option set. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of omar parihuana Sent: Friday, December 01, 2006 9:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] CALL TRANSFER Hi Guys, I'm implementing my Asterisk step by step, so far the communications between softphones, hardphones with Gateways, voice mail, are working fine. Rightnow I would like to enable Call tranfer (like Traditional PBX) in Blind Transfer and AttendXFER, I'm reading features.conf in accordance to voip-info.org http://voip-info.org/ but the transfer doesn't work! Please if you can provide me some examples will be very appreciate. Rgds. -- Omar E.P.T - Certified Networking Professionals make better Connections! http://omarept.blogspot.com/ Usysnet Corp Open Source Solutions www.usysnet.com.pe http://www.usysnet.com.pe/ ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Omar E.P.T - Certified Networking Professionals make better Connections! http://omarept.blogspot.com/ Usysnet Corp Open Source Solutions www.usysnet.com.pe ___ --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] call transfer problem
Can anyone help with the following problem please? 1) On a receptionist's phone (Snom 360 latest firmware), a call is answered. 2) While on this call a second call comes to the phone but she does not answer it. 3) The receptionist makes an attended transfer placing the first caller on hold and dialing an extension internally, but the internal party is not willing to pick up the call so she hangs up the internal call. The second call remains unanswered. 4) The receptionist now has two blinking lights on the phone for the original call and the new call is still unanswered. 5) If either button is pressed, the call that is picked up is the second call and the first call remains on hold ... anyone know why this is? The funny thing is if a blind transfer or an attended transfer that is accepted by the internal party is performed, the functions work correctly. Regards, Colin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call transfer issues
My guess is I stumped everyone ;)
Anyway, I rolled back asterisk to 1.2.9.1 (same for libpri and zaptel
back one release) and transfers were working again. Now I'm still quite
new to asterisks, I know enough to hold my own, but not enough to know
the full inter workings of it. But here is my thought:
Caller A calls in and talks to Employee B. B wants to transfer to C.
Asterisk sets up the bridge between B and C. B completes the transfer.
Now A and C are connected but there is no audio stream. If C or A puts
the other on hold, and then resumes the call, audio is restored.
By that I would say placing them on hold clears a flag or updates one to
connect the audio stream? Or am I way off on this assumption? Also if
this sounds like a possible bug, what information do I need to include,
or is good to include, when submitting bugs?
Thanks,
Kevin
Kevin Smith wrote:
Hey everyone,
Currently we are running IP601's with 1.6.6 SIP firmware on Asterisk
1.2.10. It has been reported to me when doing an attended transfer the
audio drops out. I ran a few different tests and here is what I noticed.
1. Blind transfers work with no problem.
2. Attended transfers were you transfer the call before the person
picks up works.
3. If the person the call is being transferred to answers and then the
transfer completes, the audio drops.
I noticed in the CLI the following (I replaced the number with XXX's)
-- Attempting native bridge of SIP/989XXX-b76167c8 and
SIP/989XXX-08f956b8
== Parsing '/etc/asterisk/manager.conf': Found
-- Stopped music on hold on Zap/2-1
== Spawn extension (Mercury-Directory-Dialer, 989XXX, 8) exited
non-zero on 'SIP/989XXX-b76167c8ZOMBIE'
-- Executing Hangup(SIP/989XXX-b76167c8ZOMBIE, ) in new
stack
== Spawn extension (Mercury-Directory-Dialer, h, 1) exited non-zero
on 'SIP/989XXX-b76167c8ZOMBIE'
-- Incoming call: Got SIP response 500 Internal Server Error back
from 64.7.177.103
Now what I noticed is that once the transfer is done, I'm still
connected the the person that called me to do an attended transfer.
However, if I hang up the phone, the call drops. If I place the call
on hold and take them off hold, audio is resumed and everything works
normally.
Here is the conf information
exten = s,1,SetCallerID(${ARG1})
exten = s,n,Set(DST_EXT_NUM=${ARG2})
exten = s,n,gotoif,$[${ARG2}=989XX]?TIME:GOON ;Add test if
hours is the basis for voice mail
exten = s,n(GOON),AGI(VoiceMail.php) ;Test for phone status
exten = s,n,Noop(Testing Completed||Code:${ON_PHONE}${IN_QUEUE})
exten = s,n,Dial(SIP/${ARG2},25)
...VoiceMail choice
exten = h,1,HangUp()
Where I have VoiceMail choice it takes the variables from the AGI
script and decides which voice message to play. But the problem is
happening before that occurs so I don't think it has anything to do
with the problem.
Any ideas to what could be the cause or how to correct it? SIP version
or does the new asterisk build have any new features enabled by
default that the older build would not? Any suggestions or thoughts
would be greatly helpful.
Kevin
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[asterisk-users] Call transfer issues
Hey everyone,
Currently we are running IP601's with 1.6.6 SIP firmware on Asterisk
1.2.10. It has been reported to me when doing an attended transfer the
audio drops out. I ran a few different tests and here is what I noticed.
1. Blind transfers work with no problem.
2. Attended transfers were you transfer the call before the person picks
up works.
3. If the person the call is being transferred to answers and then the
transfer completes, the audio drops.
I noticed in the CLI the following (I replaced the number with XXX's)
-- Attempting native bridge of SIP/989XXX-b76167c8 and
SIP/989XXX-08f956b8
== Parsing '/etc/asterisk/manager.conf': Found
-- Stopped music on hold on Zap/2-1
== Spawn extension (Mercury-Directory-Dialer, 989XXX, 8) exited
non-zero on 'SIP/989XXX-b76167c8ZOMBIE'
-- Executing Hangup(SIP/989XXX-b76167c8ZOMBIE, ) in new stack
== Spawn extension (Mercury-Directory-Dialer, h, 1) exited non-zero on
'SIP/989XXX-b76167c8ZOMBIE'
-- Incoming call: Got SIP response 500 Internal Server Error back
from 64.7.177.103
Now what I noticed is that once the transfer is done, I'm still
connected the the person that called me to do an attended transfer.
However, if I hang up the phone, the call drops. If I place the call on
hold and take them off hold, audio is resumed and everything works
normally.
Here is the conf information
exten = s,1,SetCallerID(${ARG1})
exten = s,n,Set(DST_EXT_NUM=${ARG2})
exten = s,n,gotoif,$[${ARG2}=989XX]?TIME:GOON ;Add test if
hours is the basis for voice mail
exten = s,n(GOON),AGI(VoiceMail.php) ;Test for phone status
exten = s,n,Noop(Testing Completed||Code:${ON_PHONE}${IN_QUEUE})
exten = s,n,Dial(SIP/${ARG2},25)
...VoiceMail choice
exten = h,1,HangUp()
Where I have VoiceMail choice it takes the variables from the AGI script
and decides which voice message to play. But the problem is happening
before that occurs so I don't think it has anything to do with the problem.
Any ideas to what could be the cause or how to correct it? SIP version
or does the new asterisk build have any new features enabled by default
that the older build would not? Any suggestions or thoughts would be
greatly helpful.
Kevin
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[asterisk-users] Call transfer asterisk + with SPA-1001
Does anybody knows how to transfer calls from Sipura SPA 1001 configured as asterisk internal ? ___ --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] Call Transfer does not work
Hi !
I am trying to transfer calls between internal SIP softclients, but it does
not work. Every time I press a key on the softclient, the CLI shows the
following output:
Attempting native bridge of SIP/456-9ee0 and SIP/173-f586
This is my extensions.conf:
[macro-voicemail]
exten = s,1,Dial(${ARG1},5,Ttr)
exten = s,2,Goto(status-${DIALSTATUS},1)
exten = status-BUSY,1,VoiceMail(b${MACRO_EXTEN})
exten = status-BUSY,2,Playback(vm-goodbye)
exten = status-BUSY,3,Hangup()
exten = status-NOANSWER,1,VoiceMail(u${MACRO_EXTEN})
exten = status-NOANSWER,2,Playback(vm-goodbye)
exten = status-NOANSWER,3,Hangup()
[internal]
exten = _ZXZ,1,Macro(voicemail,SIP/${EXTEN})
And this is the part of the features.conf I changed (just uncommented that
part)
[featuremap]
blindxfer = #1 ; Blind transfer
disconnect = *0; Disconnect
automon = *1 ; One Touch Record
atxfer = *2; Attended transfer
None of the shortcuts in [featuremap] works.
What am I doing wrong?
Regards, Jens
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[Asterisk-Users] Call Transfer Disconnect (CT-5)
Hi, Anyone has experience in using Call Transfer Disconnect (CT-5) over a PRI with Asterisk ? Call Transfer Disconnect allows you to transfer a call to a third party and disconnect yourself from the communication and also freeing your PRI channels. Here is a document that explains how it works: http://www.callamericacom.com/pdf/ctd_instructions.pdf My question is how can I do this with Asterisk, especially with a softphone (currently using SJPhone). Andre Courchesne ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call transfer to cell phone
Hi! Is anyone managed to transfer an alredy bridged call, to a cell phone? Some days ago, someone told me to look for the solution in features.conf, but I still haven't found it. I tryied to use de default blindxfer, but it only accept internal extensions. Thanks in advance, Giuseppe ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call transfer to cell phone [UPDATE]
Hi!
I tried this in features.conf
testfeature = *9,callee,Dial,CAPI/ISDN4/my_phone_number/b,60,T
and it works... but... I would be able to transfer a call to any phone
number,
so I tried to use this line:
testfeature = _*9.,callee,Dial,CAPI/ISDN4/${EXTEN:2}/b,60,T
but... Asterisk crash! (it doesn't want even to reload configuration)
Any idea about how to do so? Thanks a lot!
Giuseppe
--
In my last email I wrote:
Hi!
Is anyone managed to transfer an alredy bridged call, to a cell phone?
Some days ago, someone told me to look for the solution in
features.conf,
but I still haven't found it. I tryied to use de default blindxfer,
but it only
accept internal extensions.
Thanks in advance,
Giuseppe
--
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[Asterisk-Users] call transfer to external phone number
Hi! Is it possible to transfer a call to an external phone instead of transferring the call to internal phone? (I'm sorry for my bad english, I hope you understand) When, during a call, I digit #123, the call is transferred to internal extension 123, but if I digit #external_phone_number, it tells me that it's impossible. Any idea? Thanks a lot! Giuseppe ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call transfer to external phone number
Hi! Is it possible to transfer a call to an external phone instead of transferring the call to internal phone? (I'm sorry for my bad english, I hope you understand) When, during a call, I digit #123, the call is transferred to internal extension 123, but if I digit #external_phone_number, it tells me that it's impossible. Any idea? Thanks a lot! Giuseppe I know that with polycom I was able to do this. Not by using the # sign but by hitting the transfer button and then entering the persons number and pressing transfer. Please post your extensions.conf __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call transfer to external phone number
Yes, as long as the context that the phone transfering has an exten declared for that number. On 4/3/06, Giuseppe [EMAIL PROTECTED] wrote: Hi! Is it possible to transfer a call to an external phone instead of transferring the call to internal phone? (I'm sorry for my bad english, I hope you understand) When, during a call, I digit #123, the call is transferred to internal extension 123, but if I digit #external_phone_number, it tells me that it's impossible. Any idea? Thanks a lot! Giuseppe ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] call transfer to external phone number
From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of C F Sent: Monday, April 03, 2006 3:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] call transfer to external phone number Yes, as long as the context that the phone transfering has an exten declared for that number. Does Asterisk make any distinction between an internal number and an external number? I'm inclined to think it might be some kind of timeout issue. And I've got the proof: From my Aastra 9xxx (don't know the number and I'm to lasy to go look) I can transfer a call to any extension, including the lng extension required for dialing an external number (ie: #0X). Unfortunatelly that's the ONLY phone I can do that from! I can't do it from XLite softphone and I can't do it from analog phones connected to a Linksys PAP2. For the phones that are unable to transfer to external numbers I've got alias extensions defined (basic, 3 digit extensions). On 4/3/06, Giuseppe [EMAIL PROTECTED] wrote: Hi! Is it possible to transfer a call to an external phone instead of transferring the call to internal phone? (I'm sorry for my bad english, I hope you understand) When, during a call, I digit #123, the call is transferred to internal extension 123, but if I digit #external_phone_number, it tells me that it's impossible. Any idea? Thanks a lot! Giuseppe ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call transfer to external phone number
Asteirsk has got no clue what's internal and what's not, it's the context that decide what numbers are available for a user. In your case more info is needed to troubleshoot it. On 4/3/06, Cosmin Prund [EMAIL PROTECTED] wrote: From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of C F Sent: Monday, April 03, 2006 3:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] call transfer to external phone number Yes, as long as the context that the phone transfering has an exten declared for that number. Does Asterisk make any distinction between an internal number and an external number? I'm inclined to think it might be some kind of timeout issue. And I've got the proof: From my Aastra 9xxx (don't know the number and I'm to lasy to go look) I can transfer a call to any extension, including the lng extension required for dialing an external number (ie: #0X). Unfortunatelly that's the ONLY phone I can do that from! I can't do it from XLite softphone and I can't do it from analog phones connected to a Linksys PAP2. For the phones that are unable to transfer to external numbers I've got alias extensions defined (basic, 3 digit extensions). On 4/3/06, Giuseppe [EMAIL PROTECTED] wrote: Hi! Is it possible to transfer a call to an external phone instead of transferring the call to internal phone? (I'm sorry for my bad english, I hope you understand) When, during a call, I digit #123, the call is transferred to internal extension 123, but if I digit #external_phone_number, it tells me that it's impossible. Any idea? Thanks a lot! Giuseppe ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call transfer to external phone number
SIP transfers happen out of band, so the context is the sip phone's
context noted in sip.conf.
For Inbound and outbound (ie Dial application), the context is the entry
point in the dial plan. If you need features.conf transfers to work in
a specific context you need to set the __TRANSFER_CONTEXT variable
before the Dial application so asterisk knows what context to look for
extensions.
The relevant wiki page:
http://www.voip-info.org/wiki/view/Asterisk+config+features.conf
example:
exten = 101,1,set(__TRANSFER_CONTEXT=vm-internal)
exten = 101,n,Macro(superdial,SIP/vm-ext1SIP/outsip/9995551212,15,tr,
,pstn,2,${CALLERIDNAME},${CALLERIDNUM},pstn,[EMAIL PROTECTED])
So .. in this instance, when we outdial the cellphone (9995551212) with
the 't' option, we support transfers. If we don't set the transfer
context as above when the # key is hit. Asterisk is looking in the
[inbound] context because that is where extension 101 was dialed from.
But ... exten 101 doesn't want those available extensions, they want the
same set of extensions they have at their sip phone so they can transfer
to voicemail and so on.
Since our outbound pattern dials to SIP/outsip also exist in
[vm-internal] .. calls can be transferred out to PSTN numbers.
in any case.. this is how I got it to work. :)
Cosmin Prund wrote:
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of C F
Sent: Monday, April 03, 2006 3:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] call transfer to external phone number
Yes, as long as the context that the phone transfering has an exten
declared for that number.
Does Asterisk make any distinction between an internal number and an
external number? I'm inclined to think it might be some kind of timeout
issue. And I've got the proof:
From my Aastra 9xxx (don't know the number and I'm to lasy to go look) I can
transfer a call to any extension, including the lng extension required
for dialing an external number (ie: #0X). Unfortunatelly that's the
ONLY phone I can do that from! I can't do it from XLite softphone and I
can't do it from analog phones connected to a Linksys PAP2.
For the phones that are unable to transfer to external numbers I've got
alias extensions defined (basic, 3 digit extensions).
On 4/3/06, Giuseppe [EMAIL PROTECTED] wrote:
Hi!
Is it possible to transfer a call to an external phone instead of
transferring the call to internal phone?
(I'm sorry for my bad english, I hope you understand)
When, during a call, I digit #123, the call is transferred to internal
extension 123,
but if I digit #external_phone_number, it tells me that it's
impossible.
Any idea?
Thanks a lot!
Giuseppe
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[Asterisk-Users] Call transfer - (Call failed)
Hi,
I'm trying to call an extension and then transfer the call
to another extension, but something strange happens.
This is the extension:
exten = _9.,1,Dial(CAPI/ISDN4/${EXTEN:1}/b,tT)
When I dial any number starting with 9, I always
get CALL FAILED, but the called party still receive
the call and hears silence forever.
Does anyone know why?
Giuseppe
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[Asterisk-Users] Call transfer - (Call failed)
Hi,
I'm trying to call an extension and then transfer the call
to another extension, but something strange happens.
This is the extension:
exten = _9.,1,Dial(CAPI/ISDN4/${EXTEN:1}/b,tT)
When I dial any number starting with 9, I always
get CALL FAILED, but the called party still receive
the call and hears silence forever.
Does anyone know why?
Giuseppe
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[Asterisk-Users] Call transfer problems, SOLVED
Hi All, in regards to my previous queries about call transfers not working from
inside, several days of searching turned up this posting:
I got this to work by editing the line
exten = s,14,Dial(${OUT_${ARG1}}/${OUTNUM})
to say
exten = s,14,Dial(${OUT_${ARG1}}/${OUTNUM},,Tt)
in extensions.conf
seems like many people have had this issue in the past, I guess it's AMP
related, as I added those options in the 'general settings' dialog in AMP, but
it would never work when the call originated internally went out of the
office (oddly enough, internal calls would transfer fine)... so, no idea why
this is mucked up, but the above hint seems to have resolved my issue.
jus fyi...
Dan
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[Asterisk-Users] Call Transfer - Both legs must reside on Asterisk box to transfer at this time
I have a SIP user, 2944093 that dialled 3254102. I'm trying to transfer the call from 3254102 to 3254104. When I try and transfer the call, I get the following on the Asterisk console. Mar 3 15:14:18 NOTICE[23124]: chan_sip.c:6731 get_refer_info: Supervised transfer requested, but unable to find callid '[EMAIL PROTECTED]'. Both legs must reside on Asterisk box to transfer at this time. Below is what my SIP debug console output shows me. IP 216.188.128.11 is the phone that the transferer is on (3254102). It sends a REFER message to Asterisk. Asterisk turns around and says 'Not found' eventhough the destination user, 3254104, is in it's database. I wonder if this is because the REFER has Asterisks's IP address and not the IP address of the phone? How could it have gotten that way? Thanks, Doug. --- (10 headers 0 lines)--- -- SIP/3254104-a911 is ringing -- SIP read from 216.188.128.11:5060: REFER sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 216.188.128.11;branch=z9hG4bKb3056f7489B0729B From: sip:[EMAIL PROTECTED];tag=AD42A97D-626BB596 To: Douglas Garstang sip:[EMAIL PROTECTED];tag=as6202b08e CSeq: 2 REFER Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.3.0067 Refer-To: sip:[EMAIL PROTECTED];user=phone?Replaces=77a7b64e-f546fcbc-f206df35%40172.31.16.67%3Bto-tag%3Das4744b9fa%3Bfrom-tag%3D200C85AA-7A3B0AE3 Referred-By: sip:[EMAIL PROTECTED] Max-Forwards: 70 Content-Length: 0 --- (12 headers 0 lines)--- Transfer to 3254104 in From_OneEighty Transfer from 3254102 in From_OneEighty Mar 3 14:32:49 NOTICE[16519]: chan_sip.c:6731 get_refer_info: Supervised transfer requested, but unable to find callid '[EMAIL PROTECTED]'. Both legs must reside on Asterisk box to transfer at this time. Reliably Transmitting (no NAT) to 216.188.128.11:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 216.188.128.11;branch=z9hG4bKb3056f7489B0729B;received=216.188.128.11 From: sip:[EMAIL PROTECTED];tag=AD42A97D-626BB596 To: Douglas Garstang sip:[EMAIL PROTECTED];tag=as6202b08e Call-ID: [EMAIL PROTECTED] CSeq: 2 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Accept: application/sdp Content-Length: 0 Here's the database entry for the destination number: /SIP/Registry/3254104 : 216.188.128.12:5060:3600:3254104:sip:[EMAIL PROTECTED] As you can see, that isn't what the REFER has. It has 216.188.140.203, which is Asterisks IP address. I don't know if that's the issue or not. Asterisk _IS_ in the RTP path. Doug. -Original Message- From: David Thomas [mailto:[EMAIL PROTECTED] Sent: Friday, March 03, 2006 2:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Hardware Requirements for 1M minutes Sorry, I saw that right after I posted. It is per month. And almost all during business hours. regards, David On 3/3/06, Martin Joseph [EMAIL PROTECTED] wrote: On Mar 3, 2006, at 9:49 AM, David Thomas wrote: I'm doing an install for a client with the following requirements. - 1 Million minutes of outbound calling Per what? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Transfer - Both legs must reside on Asterisk box to transfer at this time
I have a SIP user, 2944093 that dialled 3254102. I'm trying to transfer the call from 3254102 to 3254104. When I try and transfer the call, I get the following on the Asterisk console. Mar 3 15:14:18 NOTICE[23124]: chan_sip.c:6731 get_refer_info: Supervised transfer requested, but unable to find callid '[EMAIL PROTECTED]'. Both legs must reside on Asterisk box to transfer at this time. Below is what my SIP debug console output shows me. IP 216.188.128.11 is the phone that the transferer is on (3254102). It sends a REFER message to Asterisk. Asterisk turns around and says 'Not found' eventhough the destination user, 3254104, is in it's database. I wonder if this is because the REFER has Asterisks's IP address and not the IP address of the phone? How could it have gotten that way? Thanks, Doug. --- (10 headers 0 lines)--- -- SIP/3254104-a911 is ringing -- SIP read from 216.188.128.11:5060: REFER sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 216.188.128.11;branch=z9hG4bKb3056f7489B0729B From: sip:[EMAIL PROTECTED];tag=AD42A97D-626BB596 To: Douglas Garstang sip:[EMAIL PROTECTED];tag=as6202b08e CSeq: 2 REFER Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.3.0067 Refer-To: sip:[EMAIL PROTECTED];user=phone?Replaces=77a7b64e-f546fcbc-f206df35%40172.31.16.67%3Bto-tag%3Das4744b9fa%3Bfrom-tag%3D200C85AA-7A3B0AE3 Referred-By: sip:[EMAIL PROTECTED] Max-Forwards: 70 Content-Length: 0 --- (12 headers 0 lines)--- Transfer to 3254104 in From_OneEighty Transfer from 3254102 in From_OneEighty Mar 3 14:32:49 NOTICE[16519]: chan_sip.c:6731 get_refer_info: Supervised transfer requested, but unable to find callid '[EMAIL PROTECTED]'. Both legs must reside on Asterisk box to transfer at this time. Reliably Transmitting (no NAT) to 216.188.128.11:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 216.188.128.11;branch=z9hG4bKb3056f7489B0729B;received=216.188.128.11 From: sip:[EMAIL PROTECTED];tag=AD42A97D-626BB596 To: Douglas Garstang sip:[EMAIL PROTECTED];tag=as6202b08e Call-ID: [EMAIL PROTECTED] CSeq: 2 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Accept: application/sdp Content-Length: 0 Here's the database entry for the destination number: /SIP/Registry/3254104 : 216.188.128.12:5060:3600:3254104:sip:[EMAIL PROTECTED] As you can see, that isn't what the REFER has. It has 216.188.140.203, which is Asterisks IP address. I don't know if that's the issue or not. Asterisk _IS_ in the RTP path. Doug. -Original Message- From: David Thomas [mailto:[EMAIL PROTECTED] Sent: Friday, March 03, 2006 2:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Hardware Requirements for 1M minutes Sorry, I saw that right after I posted. It is per month. And almost all during business hours. regards, David On 3/3/06, Martin Joseph [EMAIL PROTECTED] wrote: On Mar 3, 2006, at 9:49 AM, David Thomas wrote: I'm doing an install for a client with the following requirements. - 1 Million minutes of outbound calling Per what? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Transfer
Title: Call Transfer Can anyone point me in the right direction. My users (all using Sipura SPA-841 phones) need the ability to transfer a call to another number. How can I setup a dial plan to do this? David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com NEW !!! Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Transfer
I would use asterisk's built in blind or attended transfer features. This way the system is based around dtmf and the users aren't tied to a specific model of phone to accomodate future upgrades. In order to do this I would recommend editing features.conf so blindxfer = ** instead of *. A single * for transfer makes it real difficult to use banking and other IVRs that ask you to press #. Then, in each necessary Dial cmd of your dialplan, make sure there is a t or a T in the options to enable either the called or calling users to initiate the transfer. More details about the Dial command: http://www.voip-info.org/wiki-Asterisk+cmd+Dial Moj Dave Morrow wrote: Can anyone point me in the right direction. My users (all using Sipura SPA-841 phones) need the ability to transfer a call to another number. How can I setup a dial plan to do this? David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] _http://www.autodatasolutions.com_ NEW !!! Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at_ [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]_ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call transfer
I'm not sure how this is suppose to work. But I want to be able to call people from a SIP phone and transfer them into a conference room. If I call another extension that is a SIP phone I can hit # and then enter the conference room number. If I call from the PSTN to the SIP extension phone I can transfer by hitting # too. But if I call from the SIP phone extension to a PSTN number it doesn't do anything when I hit the #. I'm using [EMAIL PROTECTED] and under general settings I have tTrwW for Asterisk Dial Command Settings. Can you call through a Zap trunk from a SIP phone and do a call transfer? -- Michael Sampson Information Systems Manager Customer Contact Services [EMAIL PROTECTED] 952-936-4000 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call transfer
I got this to work by editing the line
exten = s,14,Dial(${OUT_${ARG1}}/${OUTNUM})
to say
exten = s,14,Dial(${OUT_${ARG1}}/${OUTNUM},,Tt)
in extensions.conf
Do you know of anyway to set it up through AMP, so it works with all calls?
Michael Sampson
Information Systems Manager
Customer Contact Services
[EMAIL PROTECTED]
952-936-4000
Michael Sampson wrote:
I'm not sure how this is suppose to work. But I want to be able to
call people from a SIP phone and transfer them into a conference room.
If I call another extension that is a SIP phone I can hit # and then
enter the conference room number. If I call from the PSTN to the SIP
extension phone I can transfer by hitting # too. But if I call from
the SIP phone extension to a PSTN number it doesn't do anything when I
hit the #. I'm using [EMAIL PROTECTED] and under general settings I have
tTrwW for Asterisk Dial Command Settings.
Can you call through a Zap trunk from a SIP phone and do a call transfer?
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Re: [Asterisk-Users] Call transfer with voicemail password
http://www.voip-info.org/wiki/view/Asterisk+authenticate+using+voicemail+passwords Cheers 2005/12/1, Joe Pukepail [EMAIL PROTECTED]: Look into the findme feature, there is a patch on the bug tracker to add this feature. I believe that someone shows how to do it in the dial plan. I plan on implementing this, but haven't gotten around to it yet. On 11/30/05, Benjamin Lenard [EMAIL PROTECTED] wrote: Hi, I'm trying to have an extension ring my SIP phone then try my cell phone. I can transfer the call fine to the cell but I want it to ask for a pin , voicemail pin, before transferring the call. This is so if my cell's voicemail answers , the call doesn't transfer to it. Any ideas? Thanks, Ben ___ --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 -- Giovanni Miano ___ --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] Call transfer error
When I arrived a call, I would the call transfer in to another telephone number, but Asterisk show error: Executing GotoIfTime("Zap/4-1", "08:30-12:30|mon-fri|*|*?4") in new stack -- Executing GotoIfTime("Zap/4-1", "15:30-18:30|mon-fri|*|*?4") in new stack -- Executing Goto("Zap/4-1", "6") in new stack -- Goto (isdn_incoming,0445363378,6) -- Executing Dial("Zap/4-1", "ZAP/g2/0445384225|60") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g2/0445384225 -- Zap/5-1 is proceeding passing it to Zap/4-1 -- Channel 0/2, span 2 got hangup request -- Hungup 'Zap/5-1' == No one is available to answer at this time (1:0/0/0) -- Executing Hangup("Zap/4-1", "") in new stack == Spawn exten
sion
(isdn_incoming, 0445363378, 7) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' MY ZAPATA.CONF IS: [channels] switchtype = euroisdn signalling = bri_cpe_ptmp pridialplan = local prilocaldialplan = local nationalprefix = 0 internationalprefix = 00 echocancel = yes context=isdn_incoming group = 1 channel = 1-2 group = 2 channel = 4-5 group = 3 channel = 7-8 group = 4 channel = 10-11 MY ZAPTEL.CONF is loadzone=it defaultzone=it span=1,1,3,ccs,ami span=2,2,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12 MY EXTENSIONS.CONF is [isdn_incoming] exten = 0445363378,1,GotoIfTime(${ORAMATTINO}?4) exten = 0445363378,2,GotoIfTime(${ORAPOMERIGGIO}?4) exten = 0445363378,3,Goto(6) exten =
0445363378,4,Dial(${TELEIN},60) exten = 0445363378,5,Hangup exten = 0445363378,6,Dial(ZAP/g2/0445384225,60) exten = 0445363378,7,Hangup What can I doing? Thanks
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[Asterisk-Users] Call transfer with voicemail password
Hi, I'm trying to have an extension ring my SIP phone then try my cell phone. I can transfer the call fine to the cell but I want it to ask for a pin , voicemail pin, before transferring the call. This is so if my cell's voicemail answers , the call doesn't transfer to it. Any ideas? Thanks, Ben ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call transfer with voicemail password
Look into the findme feature, there is a patch on the bug tracker to add this feature. I believe that someone shows how to do it in the dial plan. I plan on implementing this, but haven't gotten around to it yet. On 11/30/05, Benjamin Lenard [EMAIL PROTECTED] wrote: Hi,I'm trying to have an extension ring my SIP phone then try my cellphone.I can transfer the call fine to the cell but I want it to ask for a pin , voicemail pin, before transferring the call.This is so if my cell's voicemail answers , the call doesn't transferto it.Any ideas?Thanks,Ben___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call transfer with phones that cannot handle more than one line
Hi, Does anyone have a sample config for phones (like the Zyxel P2000wv2) that cannot handle more than one line. I have tried using # followed by the extension and nothing happens??? I have parking setup but for some reason we cannot retrieve the parked call. I call the user who the call is transfered to and they dial the parked extension in this case between 701 and 710 and nothing happens. I am just using the default feature file. *** features.conf [general] parkext = 700 parkpos = 701-720 context = parkedcalls parkingtime = 45 transferdigittimeout = 3 courtesytone = beep xfersound = beep xferfailsound = beeperr ;adsipark = yes findslot = next pickupexten = *8 featuredigittimeout = 500 [featuremap] blindxfer = #1 disconnect = *0 ;automon = *1 atxfer = *2 ** Thanks ___ --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] call transfer and pick chan_h323
AFAIK there were some known issues preventing call transfer from H323
terminals, at least with Innovaphone ones.
Yours
l.
On Fri, 18 Nov 2005 17:45:39 +0100, Santosh Rao
[EMAIL PROTECTED] wrote:
Hello list,
We have asterisk v1.2.0 CVS head and ooh323 in place. calls
can be made and recieved to and from extensions.
How to implement call transfer and call pickup. when using asterisk
1.0.x dtmf=inband registers and sends dtmf but with asterisk 1.2 and
ooh323 it does not.. is this a known issue ? While google heard tht
there was a issue with chan_h323.so would not send inband so tried to
install chan_0h323.so but but.. asterisk refuses to start with
chan_oh323 it says Unregistered channel type 'Modem'
my basic requirements are h323 , call pickup and call transfer? below
attached are the configurations files tht we are using currently ...
thanking for all your support ..
Extensions.conf:-
[testing]
exten = _7.,1,Pickup({66}:[EMAIL PROTECTED])
exten = 666,1,Dial(H323/192.168.1.194,100,Ttr)
exten = 667,1,Dial(H323/192.168.1.195,100,Ttr)
exten = 668,1,Dial(H323/192.168.1.196,100,Ttr)
exten = 669,1,Dial(H323/192.168.1.192,100,Ttr)
H323.conf:-
[general]
port = 1720
bindaddr = 0.0.0.0 ; this SHALL contain a single, valid IP address for
this machine
disallow=all
allow=ulaw
allow=alaw
;dtmfmode=auto
dtmfmode=inband
gatekeeper = DISABLE
context=testing
[vivek]
type=friend
host=192.168.1.194
context=testing
Callgroup=1
pickupgroup=1-9,13
[santosh]
type=friend
host=192.168.1.195
context=testing
Callgroup=1
pickupgroup=1-9,13
[binu]
type=friend
host=192.168.1.196
context=testing
Callgroup=1
pickupgroup=1-9,13
[test1]
type=friend
host=192.168.1.192
context=testing
Callgroup=1
pickupgroup=1-9,13
Features.conf:-
[general]
parkext = 700 ; What ext. to dial to park
parkpos = 701-720 ; What extensions to park calls on
context = parkedcalls ; Which context parked calls are in
pickupex = *8
[featuremap]
blindxfer = #1 ; Blind transfer
atxfer = *2 ; Attended transfer
I haven't lost my mind; it's backed up on
tape somewhere.
Santosh Rao
Trikon Electronics Pvt Ltd
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[Asterisk-Users] call transfer and pick chan_h323
Hello list,
We have asterisk v1.2.0 CVS head and ooh323 in place. calls can be
made and recieved to and from extensions.
How to implement call transfer and call pickup. when using asterisk 1.0.x
dtmf=inband registers and sends dtmf but with asterisk 1.2 and ooh323 it does
not.. is this a known issue ? While google heard tht there was a issue with
chan_h323.so would not send inband so tried to install chan_0h323.so but but..
asterisk refuses to start with chan_oh323 it says Unregistered channel type
'Modem'
my basic requirements are h323 , call pickup and call transfer? below attached
are the configurations files tht we are using currently ...
thanking for all your support ..
Extensions.conf:-
[testing]
exten = _7.,1,Pickup({66}:[EMAIL PROTECTED])
exten = 666,1,Dial(H323/192.168.1.194,100,Ttr)
exten = 667,1,Dial(H323/192.168.1.195,100,Ttr)
exten = 668,1,Dial(H323/192.168.1.196,100,Ttr)
exten = 669,1,Dial(H323/192.168.1.192,100,Ttr)
H323.conf:-
[general]
port = 1720
bindaddr = 0.0.0.0 ; this SHALL contain a single, valid IP address for this
machine
disallow=all
allow=ulaw
allow=alaw
;dtmfmode=auto
dtmfmode=inband
gatekeeper = DISABLE
context=testing
[vivek]
type=friend
host=192.168.1.194
context=testing
Callgroup=1
pickupgroup=1-9,13
[santosh]
type=friend
host=192.168.1.195
context=testing
Callgroup=1
pickupgroup=1-9,13
[binu]
type=friend
host=192.168.1.196
context=testing
Callgroup=1
pickupgroup=1-9,13
[test1]
type=friend
host=192.168.1.192
context=testing
Callgroup=1
pickupgroup=1-9,13
Features.conf:-
[general]
parkext = 700 ; What ext. to dial to park
parkpos = 701-720 ; What extensions to park calls on
context = parkedcalls ; Which context parked calls are in
pickupex = *8
[featuremap]
blindxfer = #1 ; Blind transfer
atxfer = *2 ; Attended transfer
I haven't lost my mind; it's backed up on
tape somewhere.
Santosh Rao
Trikon Electronics Pvt Ltd
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[Asterisk-Users] Call Transfer Problem with IAX2
I'm using IAX2 with VP-320I hardphones for remote users. Everything seems to be working fine except for call transfer. Is this an issue with the IAX2 itself or the phone? If I flash the same phone with SIP, the problem disappears. Regards, Shaun Singh, Manager Travelwave 1655 Dufferin Street, Suite 201 Toronto, ON M6H 3L9 Tel: (416) 652-1212 Ext 101 Fax: (416) 652-7073 Website: www.travelwave.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
Re: [Asterisk-Users] Call Transfer problems-am I missing something?
Hi,
Thanks for the clarification. I had seen that the two options
existed, but the docs for the dial() command didn't state the
difference.
On Sun, Oct 30, 2005 at 08:23:32PM -0500, David Bandel wrote:
On 10/30/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi All,
Recently got call-transfer somewhat working on my asterisk-1.0.9
install, and came across an interesting problem. I have an account on a
VOIP Provider (voipbuster using iax to be exact) and use a line like
this in extensions.conf to have it handle all outgoing calls beginning
with 1:
exten = _1NN,1,Dial(voipbuster/00${EXTEN},t)
When I call someone and press # on the phone ( I've tried this with
various softphones and a regular phone connected to a linksys pap2)
Nothing happens.However, if the called party presses # they get the
extension prompt, and can then transfer me to an other extension. Does
anyone know why the calling party can't initiate the transfer? am I
missing something?
Yes. The ,t in the Dial() options is for callee, the T is for
caller. ,tT is for both.
Ciao,
David A. Bandel
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[Asterisk-Users] Call Transfer problems-am I missing something?
Hi All,
Recently got call-transfer somewhat working on my asterisk-1.0.9
install, and came across an interesting problem. I have an account on a
VOIP Provider (voipbuster using iax to be exact) and use a line like
this in extensions.conf to have it handle all outgoing calls beginning
with 1:
exten = _1NN,1,Dial(voipbuster/00${EXTEN},t)
When I call someone and press # on the phone ( I've tried this with
various softphones and a regular phone connected to a linksys pap2)
Nothing happens.However, if the called party presses # they get the
extension prompt, and can then transfer me to an other extension. Does
anyone know why the calling party can't initiate the transfer? am I
missing something?
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Re: [Asterisk-Users] Call Transfer problems-am I missing something?
On 10/30/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi All,
Recently got call-transfer somewhat working on my asterisk-1.0.9
install, and came across an interesting problem. I have an account on a
VOIP Provider (voipbuster using iax to be exact) and use a line like
this in extensions.conf to have it handle all outgoing calls beginning
with 1:
exten = _1NN,1,Dial(voipbuster/00${EXTEN},t)
When I call someone and press # on the phone ( I've tried this with
various softphones and a regular phone connected to a linksys pap2)
Nothing happens.However, if the called party presses # they get the
extension prompt, and can then transfer me to an other extension. Does
anyone know why the calling party can't initiate the transfer? am I
missing something?
Yes. The ,t in the Dial() options is for callee, the T is for
caller. ,tT is for both.
Ciao,
David A. Bandel
--
Focus on the dream, not the competition.
- Nemesis Air Racing Team motto
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Re: [Asterisk-Users] Call Transfer problems-am I missing something?
David Bandel wrote:
On 10/30/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi All,
Recently got call-transfer somewhat working on my asterisk-1.0.9
install, and came across an interesting problem. I have an account on a
VOIP Provider (voipbuster using iax to be exact) and use a line like
this in extensions.conf to have it handle all outgoing calls beginning
with 1:
exten = _1NN,1,Dial(voipbuster/00${EXTEN},t)
When I call someone and press # on the phone ( I've tried this with
various softphones and a regular phone connected to a linksys pap2)
Nothing happens.However, if the called party presses # they get the
extension prompt, and can then transfer me to an other extension. Does
anyone know why the calling party can't initiate the transfer? am I
missing something?
Yes. The ,t in the Dial() options is for callee, the T is for
caller. ,tT is for both.
As is documented in show application dial in the Asterisk CLI.
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[Asterisk-Users] Call transfer caller ID
hello list, in my asterisk i have blind transfer and attendent transfer. when call Z which is a public call through Capi(BRI) is received by user A he can see the Caller ID of Z and if user A blind transfer the call to user B, user B can see the caller ID of user Z but when user A attendent tranfer the call to user B, user B does not get the caller ID of user Z. same timeit is notrecorded correctly in the CDR. how can solve this problem please help. best regards shaon ___ --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] Call Transfer
Hello, I have my [EMAIL PROTECTED] working beautifully for basic call function. So now I am testing extended functions for my office users and am hitting a wall. I simply need to be able to put a call on hold and forward it to any another internal extension. I have an Eezee AT-320 IAX2 phone and according to the directions, I simply select Hold enter ext hit Fwd. However when I press the button all I do is annoy the caller with loud button punching sounds. Does something need to be configured in * to allow call transfer to work? I am using an inbound trunk from Teliax- no cards, just a T1 direct to my * server. I have found transfer functions for zapatel- but as I said I am just using the T1 and have no zapatel trunks/configurations. I have also seen a lot of information for call forwarding but that sets up a permanent forward function to a specific extension. I just want to be able to say One moment, Mike can help you with that, let me transfer you and then be able to do it. Since this happens with all my AT-320 phones, I don't think it is hardware related and there is no mention of call transfer configuration for the phone itself. Thanks -R ___ --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] Call Transfer
try # and then dial the extension.On 10/20/05, Rhonda Herron [EMAIL PROTECTED] wrote: Hello,I have my [EMAIL PROTECTED] working beautifully for basic call function. So now Iam testing extended functions for my office users and am hitting a wall.I simply need to be able to put a call on hold and forward it to any another internal extension. I have an Eezee AT-320 IAX2 phone andaccording to the directions, Isimply select Hold enter ext hit Fwd.However when I press the button all I do is annoy the caller with loud button punching sounds. Does something need to be configured in * toallow call transfer to work? I am using an inbound trunk from Teliax- nocards, just a T1 direct to my * server.I have found transfer functions for zapatel- but as I said I am just using the T1 and have no zapateltrunks/configurations.I have also seen a lot of information for callforwarding but that sets up a permanent forward function to a specific extension. I just want to be able to say One moment, Mike can help youwith that, let me transfer you and then be able to do it. Since thishappens with all my AT-320 phones, I don't think it is hardware related and there is no mention of call transfer configuration for the phoneitself.Thanks-R___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephonywww.baldwintechsolutions.comPhone: 518-631-2855 x205Phone: 845-652-4578 x205 Phone: 978-203-3848 x205Fax: 518-631-2856 ___ --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] Call Transfer
I have the phone specific directions to transfer calls, but I tried your suggestion. No go. I have 3 of the Eezee phones and call transfer doesn't work on any of them, so I really don't think it is hardware related. I think the problem may be with my feature.conf which had no reference to blindxfer or atxfer. I added them so my feature.conf now looks like this: transferdigittimeout = 3 ; Number of seconds to wait between digits when transfering a call ;courtesytone = beep; Sound file to play to the parked caller ; when someone dials a parked call xfersound = beep ; to indicate an attended transfer is complete xferfailsound = beeperr; to indicate a failed transfer ;adsipark = yes ; if you want ADSI parking announcements ;pickupexten = *8 ; Configure the pickup extension. Default is *8 ;featuredigittimeout = 500 ; Max time (ms) between digits for ; feature activation. Default is 500 [featuremap] blindxfer = #; Blind transfer disconnect = *0 ; Disconnect ;automon = *1 ; One Touch Record atxfer = *2 I rebooted my * server but still no go. Are there dependencies I am not aware of? Should [featuremap] be referenced elsewhere as well? I am working with * CVS 1.0.9 and have found an article on wiki that support for call transfer was added in 1.2. Are there other places I need to hack for this functionality? Thanks, -R Tom Vile wrote: try # and then dial the extension. On 10/20/05, *Rhonda Herron* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hello, I have my [EMAIL PROTECTED] working beautifully for basic call function. So now I am testing extended functions for my office users and am hitting a wall. I simply need to be able to put a call on hold and forward it to any another internal extension. I have an Eezee AT-320 IAX2 phone and according to the directions, I simply select Hold enter ext hit Fwd. However when I press the button all I do is annoy the caller with loud button punching sounds. Does something need to be configured in * to allow call transfer to work? I am using an inbound trunk from Teliax- no cards, just a T1 direct to my * server. I have found transfer functions for zapatel- but as I said I am just using the T1 and have no zapatel trunks/configurations. I have also seen a lot of information for call forwarding but that sets up a permanent forward function to a specific extension. I just want to be able to say One moment, Mike can help you with that, let me transfer you and then be able to do it. Since this happens with all my AT-320 phones, I don't think it is hardware related and there is no mention of call transfer configuration for the phone itself. Thanks -R ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com http://www.baldwintechsolutions.com Phone: 518-631-2855 x205 Phone: 845-652-4578 x205 Phone: 978-203-3848 x205 Fax: 518-631-2856 ___ --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
Re: [Asterisk-Users] Call Transfer
maybe its not setting the DTMF tones properly. What do you have setup for the phone and extensions. Usually its rfc2833 but could be inband.On 10/20/05, Rhonda Herron [EMAIL PROTECTED] wrote: I have the phone specific directions to transfer calls, but I tried yoursuggestion. No go. I have 3 of the Eezee phones andcall transferdoesn'twork on any of them, so I really don't think it is hardware related. I think the problem may be with my feature.conf which had noreference to blindxfer or atxfer. I added them so my feature.conf nowlooks like this:transferdigittimeout = 3; Number of seconds to wait between digits when transfering a call ;courtesytone = beep; Sound file to play to the parked caller ; when someone dials a parked call xfersound = beep ; to indicate an attended transfer iscomplete xferfailsound = beeperr; to indicate a failed transfer ;adsipark = yes ; if you want ADSI parking announcements ;pickupexten = *8 ; Configure the pickup extension.Default is *8 ;featuredigittimeout = 500; Max time (ms) between digits for ; feature activation.Default is 500 [featuremap] blindxfer = #; Blind transfer disconnect = *0 ; Disconnect ;automon = *1; One Touch Record atxfer = *2I rebooted my * server but still no go. Are theredependenciesI amnot aware of? Should [featuremap] be referenced elsewhere as well? I amworking with * CVS 1.0.9 and have found an article on wiki that supportfor call transfer was added in 1.2.Are there other places I need tohack for this functionality?Thanks,-RTom Vile wrote: try # and then dial the extension. On 10/20/05, *Rhonda Herron* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hello, I have my [EMAIL PROTECTED] working beautifully for basic call function. So now I am testing extended functions for my office users and am hitting a wall. I simply need to be able to put a call on hold and forward it to any another internal extension. I have an Eezee AT-320 IAX2 phone and according to the directions, Isimply select Hold enter ext hit Fwd. However when I press the button all I do is annoy the caller with loud button punching sounds. Does something need to be configured in * to allow call transfer to work? I am using an inbound trunk from Teliax- no cards, just a T1 direct to my * server.I have found transfer functions for zapatel- but as I said I am just using the T1 and have no zapatel trunks/configurations.I have also seen a lot of information for call forwarding but that sets up a permanent forward function to a specific extension. I just want to be able to say One moment, Mike can help you with that, let me transfer you and then be able to do it. Since this happens with all my AT-320 phones, I don't think it is hardware related and there is no mention of call transfer configuration for the phone itself. Thanks -R ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com http://www.baldwintechsolutions.com Phone: 518-631-2855 x205 Phone: 845-652-4578 x205 Phone: 978-203-3848 x205 Fax: 518-631-2856 ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo 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.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephony www.baldwintechsolutions.comPhone: 518-631-2855 x205Phone: 845-652-4578 x205Phone: 978-203-3848 x205Fax: 518-631-2856 ___ --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] Call Transfer
It is set to rfc2833. Tom Vile wrote: maybe its not setting the DTMF tones properly. What do you have setup for the phone and extensions. Usually its rfc2833 but could be inband. On 10/20/05, *Rhonda Herron* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I have the phone specific directions to transfer calls, but I tried your suggestion. No go. I have 3 of the Eezee phones and call transfer doesn't work on any of them, so I really don't think it is hardware related. I think the problem may be with my feature.conf which had no reference to blindxfer or atxfer. I added them so my feature.conf now looks like this: transferdigittimeout = 3 ; Number of seconds to wait between digits when transfering a call ;courtesytone = beep; Sound file to play to the parked caller ; when someone dials a parked call xfersound = beep ; to indicate an attended transfer is complete xferfailsound = beeperr; to indicate a failed transfer ;adsipark = yes ; if you want ADSI parking announcements ;pickupexten = *8 ; Configure the pickup extension. Default is *8 ;featuredigittimeout = 500 ; Max time (ms) between digits for ; feature activation. Default is 500 [featuremap] blindxfer = #; Blind transfer disconnect = *0 ; Disconnect ;automon = *1 ; One Touch Record atxfer = *2 I rebooted my * server but still no go. Are there dependencies I am not aware of? Should [featuremap] be referenced elsewhere as well? I am working with * CVS 1.0.9 and have found an article on wiki that support for call transfer was added in 1.2. Are there other places I need to hack for this functionality? Thanks, -R Tom Vile wrote: try # and then dial the extension. On 10/20/05, *Rhonda Herron* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hello, I have my [EMAIL PROTECTED] working beautifully for basic call function. So now I am testing extended functions for my office users and am hitting a wall. I simply need to be able to put a call on hold and forward it to any another internal extension. I have an Eezee AT-320 IAX2 phone and according to the directions, I simply select Hold enter ext hit Fwd. However when I press the button all I do is annoy the caller with loud button punching sounds. Does something need to be configured in * to allow call transfer to work? I am using an inbound trunk from Teliax- no cards, just a T1 direct to my * server. I have found transfer functions for zapatel- but as I said I am just using the T1 and have no zapatel trunks/configurations. I have also seen a lot of information for call forwarding but that sets up a permanent forward function to a specific extension. I just want to be able to say One moment, Mike can help you with that, let me transfer you and then be able to do it. Since this happens with all my AT-320 phones, I don't think it is hardware related and there is no mention of call transfer configuration for the phone itself. Thanks -R ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com http://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com http://www.baldwintechsolutions.com http://www.baldwintechsolutions.com Phone: 518-631-2855 x205 Phone: 845-652-4578 x205 Phone: 978-203-3848 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com
Re: [Asterisk-Users] Call Transfer
I'm not sure the txfer functionality is in the 1.0.X branch. I'm pretty sure you will need HEAD or the 1.2 betas.On 10/20/05, Rhonda Herron [EMAIL PROTECTED] wrote:It is set to rfc2833.Tom Vile wrote: maybe its not setting the DTMF tones properly.What do you have setup for the phone and extensions.Usually its rfc2833 but could be inband. On 10/20/05, *Rhonda Herron* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I have the phone specific directions to transfer calls, but I tried your suggestion. No go. I have 3 of the Eezee phones andcall transfer doesn'twork on any of them, so I really don't think it is hardware related. I think the problem may be with my feature.conf which had no reference to blindxfer or atxfer. I added them so my feature.conf now looks like this: transferdigittimeout = 3; Number of seconds to wait between digits when transfering a call ;courtesytone = beep; Sound file to play to the parked caller; when someone dials a parked call xfersound = beep ; to indicate an attended transfer is complete xferfailsound = beeperr; to indicate a failed transfer ;adsipark = yes ; if you want ADSI parking announcements ;pickupexten = *8 ; Configure the pickup extension. Default is *8 ;featuredigittimeout = 500; Max time (ms) between digits for; feature activation.Default is 500 [featuremap] blindxfer = #; Blind transfer disconnect = *0 ; Disconnect ;automon = *1; One Touch Record atxfer = *2 I rebooted my * server but still no go. Are theredependenciesI am not aware of? Should [featuremap] be referenced elsewhere as well? I am working with * CVS 1.0.9 and have found an article on wiki that support for call transfer was added in 1.2.Are there other places I need to hack for this functionality? Thanks, -R Tom Vile wrote: try # and then dial the extension. On 10/20/05, *Rhonda Herron* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto: [EMAIL PROTECTED] wrote: Hello, I have my [EMAIL PROTECTED] working beautifully for basic call function. So now I am testing extended functions for my office users and am hitting a wall. I simply need to be able to put a call on hold and forward it to any another internal extension. I have an Eezee AT-320 IAX2 phone and according to the directions, Isimply select Hold enter ext hit Fwd. However when I press the button all I do is annoy the caller with loud button punching sounds. Does something need to be configured in * to allow call transfer to work? I am using an inbound trunk from Teliax- no cards, just a T1 direct to my * server.I have found transfer functions for zapatel- but as I said I am just using the T1 and have no zapatel trunks/configurations.I have also seen a lot of information for call forwarding but that sets up a permanent forward function to a specific extension. I just want to be able to say One moment, Mike can help you with that, let me transfer you and then be able to do it. Since this happens with all my AT-320 phones, I don't think it is hardware related and there is no mention of call transfer configuration for the phone itself. Thanks -R ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com http://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com http://www.baldwintechsolutions.com http://www.baldwintechsolutions.com Phone: 518-631-2855 x205 Phone: 845-652-4578 x205 Phone: 978-203-3848 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto: 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 http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto: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 -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design
Re: [Asterisk-Users] Call Transfer
Blind transfer should work fine #. Can you dial into Voicemail and enter your password succesfully?On 10/20/05, BJ Weschke [EMAIL PROTECTED] wrote:I'm not sure the txfer functionality is in the 1.0.X branch. I'm pretty sure you will need HEAD or the 1.2 betas.On 10/20/05, Rhonda Herron [EMAIL PROTECTED] wrote: It is set to rfc2833.Tom Vile wrote: maybe its not setting the DTMF tones properly.What do you have setup for the phone and extensions.Usually its rfc2833 but could be inband. On 10/20/05, *Rhonda Herron* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I have the phone specific directions to transfer calls, but I tried your suggestion. No go. I have 3 of the Eezee phones andcall transfer doesn'twork on any of them, so I really don't think it is hardware related. I think the problem may be with my feature.conf which had no reference to blindxfer or atxfer. I added them so my feature.conf now looks like this: transferdigittimeout = 3; Number of seconds to wait between digits when transfering a call ;courtesytone = beep; Sound file to play to the parked caller; when someone dials a parked call xfersound = beep ; to indicate an attended transfer is complete xferfailsound = beeperr; to indicate a failed transfer ;adsipark = yes ; if you want ADSI parking announcements ;pickupexten = *8 ; Configure the pickup extension. Default is *8 ;featuredigittimeout = 500; Max time (ms) between digits for; feature activation.Default is 500 [featuremap] blindxfer = #; Blind transfer disconnect = *0 ; Disconnect ;automon = *1; One Touch Record atxfer = *2 I rebooted my * server but still no go. Are theredependenciesI am not aware of? Should [featuremap] be referenced elsewhere as well? I am working with * CVS 1.0.9 and have found an article on wiki that support for call transfer was added in 1.2.Are there other places I need to hack for this functionality? Thanks, -R Tom Vile wrote: try # and then dial the extension. On 10/20/05, *Rhonda Herron* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto: [EMAIL PROTECTED] mailto: [EMAIL PROTECTED] wrote: Hello, I have my [EMAIL PROTECTED] working beautifully for basic call function. So now I am testing extended functions for my office users and am hitting a wall. I simply need to be able to put a call on hold and forward it to any another internal extension. I have an Eezee AT-320 IAX2 phone and according to the directions, Isimply select Hold enter ext hit Fwd. However when I press the button all I do is annoy the caller with loud button punching sounds. Does something need to be configured in * to allow call transfer to work? I am using an inbound trunk from Teliax- no cards, just a T1 direct to my * server.I have found transfer functions for zapatel- but as I said I am just using the T1 and have no zapatel trunks/configurations.I have also seen a lot of information for call forwarding but that sets up a permanent forward function to a specific extension. I just want to be able to say One moment, Mike can help you with that, let me transfer you and then be able to do it. Since this happens with all my AT-320 phones, I don't think it is hardware related and there is no mention of call transfer configuration for the phone itself. Thanks -R ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com http://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto: Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com mailto: Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com http://www.baldwintechsolutions.com http://www.baldwintechsolutions.com Phone: 518-631-2855 x205 Phone: 845-652-4578 x205 Phone: 978-203-3848 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto: 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 http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com
Re: [Asterisk-Users] Call Transfer
Yes, I can dial *97 for VM and check messages. When I select # during a call it does nothing though. I tried inband for DTMF but that didnt work. Am going to run debug mode ( first I have to figure out how :) ) and I will let you know what I find out. Thanks so far, R Tom Vile wrote: Blind transfer should work fine #. Can you dial into Voicemail and enter your password succesfully? On 10/20/05, *BJ Weschke* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I'm not sure the txfer functionality is in the 1.0.X branch. I'm pretty sure you will need HEAD or the 1.2 betas. On 10/20/05, *Rhonda Herron* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: It is set to rfc2833. Tom Vile wrote: maybe its not setting the DTMF tones properly. What do you have setup for the phone and extensions. Usually its rfc2833 but could be inband. On 10/20/05, *Rhonda Herron* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I have the phone specific directions to transfer calls, but I tried your suggestion. No go. I have 3 of the Eezee phones and call transfer doesn't work on any of them, so I really don't think it is hardware related. I think the problem may be with my feature.conf which had no reference to blindxfer or atxfer. I added them so my feature.conf now looks like this: transferdigittimeout = 3 ; Number of seconds to wait between digits when transfering a call ;courtesytone = beep; Sound file to play to the parked caller ; when someone dials a parked call xfersound = beep ; to indicate an attended transfer is complete xferfailsound = beeperr; to indicate a failed transfer ;adsipark = yes ; if you want ADSI parking announcements ;pickupexten = *8 ; Configure the pickup extension. Default is *8 ;featuredigittimeout = 500 ; Max time (ms) between digits for ; feature activation. Default is 500 [featuremap] blindxfer = #; Blind transfer disconnect = *0 ; Disconnect ;automon = *1 ; One Touch Record atxfer = *2 I rebooted my * server but still no go. Are there dependencies I am not aware of? Should [featuremap] be referenced elsewhere as well? I am working with * CVS 1.0.9 and have found an article on wiki that support for call transfer was added in 1.2. Are there other places I need to hack for this functionality? Thanks, -R Tom Vile wrote: try # and then dial the extension. On 10/20/05, *Rhonda Herron* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hello, I have my [EMAIL PROTECTED] working beautifully for basic call function. So now I am testing extended functions for my office users and am hitting a wall. I simply need to be able to put a call on hold and forward it to any another internal extension. I have an Eezee AT-320 IAX2 phone and according to the directions, I simply select Hold enter ext hit Fwd. However when I press the button all I do is annoy the caller with loud button punching sounds. Does something need to be configured in * to allow call transfer to work? I am using an inbound trunk from Teliax- no cards, just a T1 direct to my * server. I have found transfer functions for zapatel- but as I said I am just using the T1 and have no zapatel trunks/configurations. I have also seen a lot of information for call forwarding but that sets up a permanent forward function to a specific extension. I just want to be able to say One moment, Mike can help you with that, let me transfer you and then be able to do it. Since this happens with all my AT-320 phones, I don't think it is hardware related and there is no mention of call transfer configuration for the phone itself. Thanks -R ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com
Re: [Asterisk-Users] Call Transfer
He is what happens from the time the extension is selected from the time
the digital receptionist answers until I hangup. I watched the logs as
I was pushing all sorts of transfer button possibilities and nothing. It
just stayed at 'ooh, voice format changed to 4' Which, while humorous
tells me nothing except that my phone is not able to communicate with
the sever at all from the time the call is put through until the call is
done.
Oct 20 15:33:45 VERBOSE[2909]: -- Executing
Dial(IAX2/[EMAIL PROTECTED]/4, IAX2/7878|15|tr) in new stack
Oct 20 15:33:45 DEBUG[2909]: SIMPLE DIAL (NO URL)
Oct 20 15:33:45 VERBOSE[2909]: -- Called 7878
Oct 20 15:33:45 VERBOSE[2909]: -- Call accepted by xxx.xxx.xxx.xxx
(format ulaw)
Oct 20 15:33:45 VERBOSE[2909]: -- Format for call is ulaw
Oct 20 15:33:45 VERBOSE[2909]: -- IAX2/7878/8 is ringing
Oct 20 15:33:50 VERBOSE[2909]: -- IAX2/7878/8 answered
IAX2/[EMAIL PROTECTED]/4
Oct 20 15:33:50 VERBOSE[2909]: -- Attempting native bridge of
IAX2/[EMAIL PROTECTED]/4 and IAX2/7878/8
Oct 20 15:33:50 DEBUG[2909]: Ooh, voice format changed to 4
Here is the extension config for 7878:
exten = 7878,1,Macro(exten-vm,[EMAIL PROTECTED],7878)
And this is the config for aah_1 ( our digital receptionist)
[aa_1]
include = aa_1-custom
exten = 1,1,Goto(,s,1);
exten = fax,1,Goto(ext-fax,in_fax,1);
exten = h,1,Hangup();
exten = i,1,Playback(invalid);
exten = i,2,Goto(s,7);
include = ext-local
include = app-messagecenter
include = app-directory
exten = s,1,GotoIf($[${DIALSTATUS} = ANSWER]?4);
exten = s,2,Answer();
exten = s,3,Wait(1);
exten = s,4,SetVar(DIR-CONTEXT=default);
exten = s,5,DigitTimeout(3); Basic
exten = s,6,ResponseTimeout(7);
exten = s,7,Background(custom/aa_1);
Thanks, once again,
-R
Rhonda Herron wrote:
Yes, I can dial *97 for VM and check messages. When I select # during
a call it does nothing though. I tried inband for DTMF but that didnt
work. Am going to run debug mode ( first I have to figure out how :) )
and I will let you know what I find out.
Thanks so far,
R
Tom Vile wrote:
Blind transfer should work fine #. Can you dial into Voicemail and
enter your password succesfully?
On 10/20/05, *BJ Weschke* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I'm not sure the txfer functionality is in the 1.0.X branch. I'm
pretty sure you will need HEAD or the 1.2 betas.
On 10/20/05, *Rhonda Herron* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
It is set to rfc2833.
Tom Vile wrote:
maybe its not setting the DTMF tones properly. What do you
have setup
for the phone and extensions. Usually its rfc2833 but could
be inband.
On 10/20/05, *Rhonda Herron* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
I have the phone specific directions to transfer calls,
but I
tried your
suggestion. No go. I have 3 of the Eezee phones and call
transfer
doesn't work on any of them, so I really don't think it
is hardware
related. I think the problem may be with my feature.conf
which had no
reference to blindxfer or atxfer. I added them so my
feature.conf now
looks like this:
transferdigittimeout = 3 ; Number of seconds to
wait between
digits when transfering a call
;courtesytone = beep; Sound file to play to
the parked
caller
; when someone dials a
parked call
xfersound = beep ; to indicate an attended
transfer is
complete
xferfailsound = beeperr; to indicate a failed
transfer
;adsipark = yes ; if you want ADSI parking
announcements
;pickupexten = *8 ; Configure the pickup
extension.
Default is *8
;featuredigittimeout = 500 ; Max time (ms) between
digits for
; feature
activation. Default is 500
[featuremap]
blindxfer = #; Blind transfer
disconnect = *0 ; Disconnect
;automon = *1 ; One Touch Record
atxfer = *2
I rebooted my * server but still no go. Are
there dependencies I am
not aware of? Should [featuremap] be referenced elsewhere
as well?
I am
working with * CVS 1.0.9 and have found an article on
wiki that
support
for call transfer was added in 1.2. Are there other
places I need to
hack for this functionality?
Thanks,
-R
Tom Vile wrote:
try # and then dial the extension.
On 10/20/05, *Rhonda Herron* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
mailto: [EMAIL PROTECTED]
[Asterisk-Users] Call transfer - atxfer
Hi, I try to set up attended transfer in my Asterisk Box . My features.conf look like this: [general] parkext = 100 parkpos = 1-5 context = parkedcalls parkingtime = 100 transferdigittimeout = 3l courtesytone = beep xfersound = beep xferfailsound = invalid featuredigittimeout = 500 ;adsipark = yes pickupexten = *8 [featuremap] atxfer = *2 blindxfer = # disconnect = *0 automon = *1 and when I press *2 console says something like this: Oct 17 15:52:23 DEBUG[20916]: rtp.c:190 send_dtmf: Sending dtmf: 42 (*), at 10.2.20.65 Oct 17 15:52:23 DEBUG[20916]: channel.c:2762 ast_channel_bridge: Got AST_BRIDGE_DTMF_CHANNEL_1 on c1 (SIP/rafal-89b1) Oct 17 15:52:23 DEBUG[20916]: channel.c:2798 ast_channel_bridge: Bridge stops bridging channels SIP/andrzej-0265 and SIP/rafal-89b1 Oct 17 15:52:23 DEBUG[20916]: res_features.c:600 ast_bridge_call: Read from SIP/rafal-89b1 (1,42) -- Attempting native bridge of SIP/andrzej-0265 and SIP/rafal-89b1 Oct 17 15:52:23 DEBUG[20916]: rtp.c:190 send_dtmf: Sending dtmf: 50 (2), at 10.2.20.65 Oct 17 15:52:23 DEBUG[20916]: channel.c:2762 ast_channel_bridge: Got AST_BRIDGE_DTMF_CHANNEL_1 on c1 (SIP/rafal-89b1) Oct 17 15:52:23 DEBUG[20916]: channel.c:2798 ast_channel_bridge: Bridge stops bridging channels SIP/andrzej-0265 and SIP/rafal-89b1 Oct 17 15:52:23 DEBUG[20916]: res_features.c:600 ast_bridge_call: Read from SIP/rafal-89b1 (1,50) -- Attempting native bridge of SIP/andrzej-0265 and SIP/rafal-89b1 Does anyone know what's going on? What should I do to make attended transfer works well? Cheers Andrew ___ --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] Call transfer - atxfer
Are you using 1.0.x? DTMF Attended Transfer is not supported in 1.0.x. Unless you have a brain dead phone, you should be able to use SIP attended transfer in 1.0.x. (that would be the transfer key on the phone) Andrew Nowrot wrote: Hi, I try to set up attended transfer in my Asterisk Box . My features.conf look like this: [general] parkext = 100 parkpos = 1-5 context = parkedcalls parkingtime = 100 transferdigittimeout = 3l courtesytone = beep xfersound = beep xferfailsound = invalid featuredigittimeout = 500 ;adsipark = yes pickupexten = *8 [featuremap] atxfer = *2 blindxfer = # disconnect = *0 automon = *1 and when I press *2 console says something like this: Oct 17 15:52:23 DEBUG[20916]: rtp.c:190 send_dtmf: Sending dtmf: 42 (*), at 10.2.20.65 Oct 17 15:52:23 DEBUG[20916]: channel.c:2762 ast_channel_bridge: Got AST_BRIDGE_DTMF_CHANNEL_1 on c1 (SIP/rafal-89b1) Oct 17 15:52:23 DEBUG[20916]: channel.c:2798 ast_channel_bridge: Bridge stops bridging channels SIP/andrzej-0265 and SIP/rafal-89b1 Oct 17 15:52:23 DEBUG[20916]: res_features.c:600 ast_bridge_call: Read from SIP/rafal-89b1 (1,42) -- Attempting native bridge of SIP/andrzej-0265 and SIP/rafal-89b1 Oct 17 15:52:23 DEBUG[20916]: rtp.c:190 send_dtmf: Sending dtmf: 50 (2), at 10.2.20.65 Oct 17 15:52:23 DEBUG[20916]: channel.c:2762 ast_channel_bridge: Got AST_BRIDGE_DTMF_CHANNEL_1 on c1 (SIP/rafal-89b1) Oct 17 15:52:23 DEBUG[20916]: channel.c:2798 ast_channel_bridge: Bridge stops bridging channels SIP/andrzej-0265 and SIP/rafal-89b1 Oct 17 15:52:23 DEBUG[20916]: res_features.c:600 ast_bridge_call: Read from SIP/rafal-89b1 (1,50) -- Attempting native bridge of SIP/andrzej-0265 and SIP/rafal-89b1 Does anyone know what's going on? What should I do to make attended transfer works well? ___ --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] Call transfer - atxfer
Hi, Thank for the Email I'm using 1.0.9 so probably I'm will not have this feature. In which version of Asterisk the DTMF Attended Transfer is supported, in 1.2 Beta? Best wishes Andrew ___ --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] Call transfer - atxfer
Andrew Nowrot wrote: Hi, Thank for the Email I'm using 1.0.9 so probably I'm will not have this feature. In which version of Asterisk the DTMF Attended Transfer is supported, in 1.2 Beta? CVS-HEAD and 1.2Beta1 and later. ___ --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] Call transfer.
Hello, how i can tranfer call to another user? Im using X-Lite, i have configured in features.conf: [featuremap] blindxfer = #1 disconnect = *0 automon = *1 atxfer = *2 But when im dial *2 in conversation nothig happens. What can br problem? Im using asterisk CVS-HEAD from 02/09/05. Regards, Adam Rybak ___ --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] call transfer problem - something strange
Hi, I try to set up planet VIP-050 with asterisk. Everything works fine instead of the call transfer. When I press # console says something like this: Oct 5 11:11:20 DEBUG[25104]: chan_sip.c: sip_rtp_read: Oooh, format changed to 1024 Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 DEBUG[25104]: rtp.c:1193 ast_rtp_write: Ooh, format changed from ulaw to ilbc Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]:
Re: [SPAM:***** SpamScore] Re: [SPAM:******** SpamScore] [Asterisk-Users] Call Transfer using SIP clients
I'm having the same problem you had Frank, so I'm pleased you came up with a fix. No luck for me yet! Incoming outgoing calls work fine using X-Lite, I just cannot transfer. It's the first time I've ventured in to features.conf so I'm likely doing something silly. I'd be grateful if you could have a look. I've posted (parts of) my sip and features .conf files below. Do I need something special in extensions.conf? What's supposed to happen when I dial *1, do I hear a special dialtone and then enter the extension? Thanks, Hugh ** features.conf (mostly default): ; Sample Parking configuration ; [general] parkext = 700 ; What ext. to dial to park parkpos = 701-720 ; What extensions to park calls on context = parkedcalls ; Which context parked calls are in ;parkingtime = 45 ; Number of seconds a call can be parked for ; (default is 45 seconds) transferdigittimeout = 3 ; Number of seconds to wait between digits when transfering a call courtesytone = beep ; Sound file to play to the parked caller ; when someone dials a parked call xfersound = beep; to indicate an attended transfer is complete xferfailsound = beeperr ; to indicate a failed transfer ;adsipark = yes ; if you want ADSI parking announcements ;findslot = next ; Continue to the 'next' parking space. Defaults to 'first' available ;pickupexten = *8 ; Configure the pickup extension. Default is *8 featuredigittimeout = 500 ; Max time (ms) between digits for ; feature activation. Default is 500 [featuremap] blindxfer = *1 ; Blind transfer ;disconnect = *0 ; Disconnect ;automon = *1 ; One Touch Record atxfer = *2; Attended transfer ** sip.conf : [301] ; My phone ; type=friend ; friend means this device takes and makes calls username=301 ; Username on device callerid=Me999-999- secret= ; Password for device nat=no host=dynamic ; This host is not on the same IP addr every time context=internal ; Inbound calls from this host go here mailbox=301 ; Activate the message waiting light if this ; voicemail box has messages in it canreinvite=no; Leave this alone for now; see archives for details disallow=all allow=gsm allow=ulaw allow=alaw On 7/5/05, Frank Schoep [EMAIL PROTECTED] wrote: On Tuesday 05 July 2005 09:29, Frank Schoep wrote: If I find out how to get it working, I will append that information to the thread so others can reuse that knowledge later on, I'm sure someone will appreciate it. So, I just got X-Lite working alongside Asterisk, the problem was (call it a premonition) the fact that I set them up to send DTMFs in band. Setting this option to disabled made the X-Lite softphone work flawlessly. I hope that helps someone. Sincerely, Frank ___ 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
