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 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 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 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
[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
[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]:
[Asterisk-Users] Call Transfer Problem
I do not want to use the default key of '#' for call transfer, because as we all know, it interferes with many IVRs that require # as a termination character. I modified features.conf and added: [featuremap] atxfer = ** The double-star now works great. If I press it while on a call, I go into transfer mode. The problem is that the # still works as well! Shouldn't the atzfer specification turn off the #? Any insight would be appreciated. Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ 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 Problem
Adam Robins wrote: The double-star now works great. If I press it while on a call, I go into transfer mode. The problem is that the # still works as well! Shouldn't the atzfer specification turn off the #? Blind transfers are on '#' by default, so you may need to move them to another sequence as well. ___ 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 Problem
On Fri, 8 Oct 2004, Michael Nolan wrote: Hi ! I have checked my asterisk. It contains this patch or thBis patch is already compiled into it. can you plz guide me as to how i can make use of it ? I have pressed '#' but it doesnot give me any dial tone. Are there any special changes that need to be done in extensions.conf to make it work ? plz help me in this regard. Usman. This patch works a treat for us: http://bugs.digium.com/bug_view_page.php?bug_id=0002460 Makes all # transfers attended, but the transfer button on the phones can still be used for blind transfers from our SIP phones. Cheers, Michael On Fri, 8 Oct 2004 01:56:53 -0500 (CDT), [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi Users, I am having a prblem using attended call transfer with asterisk. Actually my sip phone does not seem to support it. Can i use attended call transfer using the dial plan ... ??? means can somehow messing up with extesnions.conf I can get attended call transfer ? And yes also is there any way I can do it with AGI scripting ? Any AGI similar examples will be a lot of help. Thanks ! Usman. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Transfer Problem
you need the x or X option to your Dial command. show application dial is your friend ... cheers Michael On Mon, 11 Oct 2004 08:37:36 -0500 (CDT), [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: On Fri, 8 Oct 2004, Michael Nolan wrote: Hi ! I have checked my asterisk. It contains this patch or thBis patch is already compiled into it. can you plz guide me as to how i can make use of it ? I have pressed '#' but it doesnot give me any dial tone. Are there any special changes that need to be done in extensions.conf to make it work ? plz help me in this regard. Usman. This patch works a treat for us: http://bugs.digium.com/bug_view_page.php?bug_id=0002460 Makes all # transfers attended, but the transfer button on the phones can still be used for blind transfers from our SIP phones. Cheers, Michael On Fri, 8 Oct 2004 01:56:53 -0500 (CDT), [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi Users, I am having a prblem using attended call transfer with asterisk. Actually my sip phone does not seem to support it. Can i use attended call transfer using the dial plan ... ??? means can somehow messing up with extesnions.conf I can get attended call transfer ? And yes also is there any way I can do it with AGI scripting ? Any AGI similar examples will be a lot of help. Thanks ! Usman. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Bielicki ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call transfer problem
My dial statement is (for testing purposes): 123,1,Dial(H323/192.168.1.55|20|tT) When a caller dials extension 123 I can connect and talk without difficulty. Both the caller and the callee can press # to drop back to asterisk. The caller can dial an extension and transfer the callee. When the callee tries to dial an extension, I get: Unable to find extension 'first digit or two' in context ' ' It seems to me that the callee is not given a proper context and therefore cannot dial extensions in asterisk without first calling the pbx. If the pbx calls an extension, that extension is in limbo. ManxPower from the Asterisk IRC has had the same problem but has not needed the transfer capability so has never looked into it farther. I know other people have gotten this to work because I've read testimonys in the mailing list archives saying that they did get it working. I'm wondering how this was accomplished? I am using a week old version of asterisk from cvs on an Athlon 600 with 512 Megs of RAM, Slackware 8.1, 2.4.20 kernel. John Fortman. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Transfer problem
For testing purposes, my dial line is:
Dial(${ARG2},20,tT)
When I call from one machine through asterisk to
another, I can press # from either side and hear "Transfer."
However, from the caller side I can continue on and
put people on hold by dialing '700'.
From the callee side, I can press # but if I try to
dial an extension I hear "I'm sorry. That is not a valid extension.
Please try again." Asterisk displays a message "Unable to find extension
'7' in context ' ' "
What this tells me is that if a VOIP client picks
up a line that has been Dial()ed from asterisk, that client is not given a
context and, therefore, cannot dial extensions. How can this be
fixed? Have I messed up the setup somehow? If so, can anyone give me
a working example?
John.
Re: [Asterisk-Users] Call Transfer Problem
yes, u are quite right, you can find this feature in almost every pbx now. We are also wondering whether, presently some one is implementing this feature or not, if no body is doing that, we can start on that Surajee - Original Message - From: George Lin To: [EMAIL PROTECTED] Sent: Wednesday, June 04, 2003 3:36 AM Subject: RE: [Asterisk-Users] Call Transfer Problem so, What should the call initiator do if s/he wants to transfer the call initiated by himself/herself, by using flash keypad or what else ? I can see such application can be used in some big office, where the BOSS always asks the secretary to make the call, once the call is connected, then the secretary can trasfer the call to the BOSS. in order to let the BOSS talk on the phone. am I right ?? Please let me know once the feature is implemented. George Lin -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Surajee RatnayakeSent: Monday, June 02, 2003 1:05 AMTo: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Call Transfer Problem U get the following output when u execute the "show application Dial" command in the Asterisk prompt, -= Info about application 'Dial' =- [Synopsis]: Place an call and connect to the current channel [Description]: Dial(Technology/resource[Technology2/resource2...][|timeout][|options][|URL]):Requests one or more channels and places specified outgoing calls on them.As soon as a channel answers, the Dial app will answer the originatingchannel (if it needs to be answered) and will bridge a call with the channelwhich first answered. All other calls placed by the Dial app will be hunp upf a timeout is not specified, the Dial application will wait indefinitelyuntil either one of the called channels answers, the user hangs up, or allchannels return busy or error. In general, the dialler will return 0 if itwas unable to place the call, or the timeout expired. However, if allchannels were busy, and there exists an extension with priority n+101 (wheren is the priority of the dialler instance), then it will be the nextexecuted extension (this allows you to setup different behavior on busy fromno-answer). This application returns -1 if the originating channel hangs up, or if thecall is bridged and either of the parties in the bridge terminate the call.The option string may contain zero or more of the following characters: 't' -- allow the called user transfer the calling user 'T' -- to allow the calling user to transfer the call. 'r' -- indicate ringing to the calling party, pass no audio until answered. 'm' -- provide hold music to the calling party until answered. 'd' -- data-quality (modem) call (minimum delay). 'c' -- clear-channel data call (PRI-PRI only). 'H' -- allow caller to hang up by hitting *. 'C' -- reset call detail record for this call. 'P[(x)]' -- privacy mode, using 'x' as database if provided. In addition to transferring the call, a call may be parked and then pickedup by another user. The optionnal URL will be sent to the called party if the channel supportsit. Surajee - Original Message - From: George Lin To: [EMAIL PROTECTED] Sent: Monday, June 02, 2003 1:11 PM Subject: FW: [Asterisk-Users] Call Transfer Problem Hi, Which document describes the Dial with T option ? Could you let me know or email it to me. Thanks, George Lin -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Surajee RatnayakeSent: Sunday, June 01, 2003 9:10 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Call Transfer Problem hi All, We are working on Soft-PBX using Asterisk. This relates to CALL TRANSFERRING aspects of Asterisk. We were able to do one type of call transfering, ie, the called person can transfer the original call to another person. but we were unable to do the other, that is, call initiator him/her self couldn't transfer the call. Eventhough the documentation for Dial applicationintructs to use "T" to achieve that. and we learnt that it has not been implemented yet in Asterisk. Is this true? Is some one workin on this issue? if the answer is NO, we can give a try to implement it, with a help of u all , ofcourse :-) (cos, we are quite new to asterisk-only 1 month of experience, but amazed
Re: [Asterisk-Users] Call Transfer Problem
Sorry, I might be being stupid, but I don't see what the problem is. Following your example, 1. Secretary calls someone for the Boss 2. Other caller answers, Secretary asks other end to wait. 3. Secretary presses the flash button (or recall or whatever it's called on the phone) 4. Secretary dial boss, tells boss that caller is on the line 5. Secretary hangs up, boss has caller. Andy *** REPLY SEPARATOR *** On 04/06/2003 at 16:11 Surajee Ratnayake wrote: yes, u are quite right, you can find this feature in almost every pbx now. We are also wondering whether, presently some one is implementing this feature or not, if no body is doing that, we can start on that Surajee - Original Message - From: George Lin To: [EMAIL PROTECTED] Sent: Wednesday, June 04, 2003 3:36 AM Subject: RE: [Asterisk-Users] Call Transfer Problem so, What should the call initiator do if s/he wants to transfer the call initiated by himself/herself, by using flash keypad or what else ? I can see such application can be used in some big office, where the BOSS always asks the secretary to make the call, once the call is connected, then the secretary can trasfer the call to the BOSS. in order to let the BOSS talk on the phone. am I right ?? Please let me know once the feature is implemented. George Lin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Surajee Ratnayake Sent: Monday, June 02, 2003 1:05 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Call Transfer Problem U get the following output when u execute the show application Dial command in the Asterisk prompt, -= Info about application 'Dial' =- [Synopsis]: Place an call and connect to the current channel [Description]: Dial(Technology/resource[Technology2/resource2...][|timeout][|options][|URL]): Requests one or more channels and places specified outgoing calls on them. As soon as a channel answers, the Dial app will answer the originating channel (if it needs to be answered) and will bridge a call with the channel which first answered. All other calls placed by the Dial app will be hunp up f a timeout is not specified, the Dial application will wait indefinitely until either one of the called channels answers, the user hangs up, or all channels return busy or error. In general, the dialler will return 0 if it was unable to place the call, or the timeout expired. However, if all channels were busy, and there exists an extension with priority n+101 (where n is the priority of the dialler instance), then it will be the next executed extension (this allows you to setup different behavior on busy from no-answer). This application returns -1 if the originating channel hangs up, or if the call is bridged and either of the parties in the bridge terminate the call. The option string may contain zero or more of the following characters: 't' -- allow the called user transfer the calling user 'T' -- to allow the calling user to transfer the call. 'r' -- indicate ringing to the calling party, pass no audio until answered. 'm' -- provide hold music to the calling party until answered. 'd' -- data-quality (modem) call (minimum delay). 'c' -- clear-channel data call (PRI-PRI only). 'H' -- allow caller to hang up by hitting *. 'C' -- reset call detail record for this call. 'P[(x)]' -- privacy mode, using 'x' as database if provided. In addition to transferring the call, a call may be parked and then picked up by another user. The optionnal URL will be sent to the called party if the channel supports it. Surajee - Original Message - From: George Lin To: [EMAIL PROTECTED] Sent: Monday, June 02, 2003 1:11 PM Subject: FW: [Asterisk-Users] Call Transfer Problem Hi, Which document describes the Dial with T option ? Could you let me know or email it to me. Thanks, George Lin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Surajee Ratnayake Sent: Sunday, June 01, 2003 9:10 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Call Transfer Problem hi All, We are working on Soft-PBX using Asterisk. This relates to CALL TRANSFERRING aspects of Asterisk. We were able to do one type of call transfering, ie, the called person can transfer the original call to another person. but we were unable to do the other, that is, call initiator him/her self couldn't transfer the call. Eventhough the documentation for Dial application intructs to use T to achieve that. and we learnt that it has
[Asterisk-Users] Call Transfer Problem
hi All, We are working on Soft-PBX using Asterisk. This relates to CALL TRANSFERRING aspects of Asterisk. We were able to do one type of call transfering, ie, the called person can transfer the original call to another person. but we were unable to do the other, that is, call initiator him/her self couldn't transfer the call. Eventhough the documentation for Dial applicationintructs to use "T" to achieve that. and we learnt that it has not been implemented yet in Asterisk. Is this true? Is some one workin on this issue? if the answer is NO, we can give a try to implement it, with a help of u all , ofcourse :-) (cos, we are quite new to asterisk-only 1 month of experience, but amazed of its great performance) Thank you very much, Surajee
Re: [Asterisk-Users] Call Transfer Problem
U get the following output when u execute the "show application Dial" command in the Asterisk prompt, -= Info about application 'Dial' =- [Synopsis]: Place an call and connect to the current channel [Description]: Dial(Technology/resource[Technology2/resource2...][|timeout][|options][|URL]):Requests one or more channels and places specified outgoing calls on them.As soon as a channel answers, the Dial app will answer the originatingchannel (if it needs to be answered) and will bridge a call with the channelwhich first answered. All other calls placed by the Dial app will be hunp upf a timeout is not specified, the Dial application will wait indefinitelyuntil either one of the called channels answers, the user hangs up, or allchannels return busy or error. In general, the dialler will return 0 if itwas unable to place the call, or the timeout expired. However, if allchannels were busy, and there exists an extension with priority n+101 (wheren is the priority of the dialler instance), then it will be the nextexecuted extension (this allows you to setup different behavior on busy fromno-answer). This application returns -1 if the originating channel hangs up, or if thecall is bridged and either of the parties in the bridge terminate the call.The option string may contain zero or more of the following characters: 't' -- allow the called user transfer the calling user 'T' -- to allow the calling user to transfer the call. 'r' -- indicate ringing to the calling party, pass no audio until answered. 'm' -- provide hold music to the calling party until answered. 'd' -- data-quality (modem) call (minimum delay). 'c' -- clear-channel data call (PRI-PRI only). 'H' -- allow caller to hang up by hitting *. 'C' -- reset call detail record for this call. 'P[(x)]' -- privacy mode, using 'x' as database if provided. In addition to transferring the call, a call may be parked and then pickedup by another user. The optionnal URL will be sent to the called party if the channel supportsit. Surajee - Original Message - From: George Lin To: [EMAIL PROTECTED] Sent: Monday, June 02, 2003 1:11 PM Subject: FW: [Asterisk-Users] Call Transfer Problem Hi, Which document describes the Dial with T option ? Could you let me know or email it to me. Thanks, George Lin -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Surajee RatnayakeSent: Sunday, June 01, 2003 9:10 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Call Transfer Problem hi All, We are working on Soft-PBX using Asterisk. This relates to CALL TRANSFERRING aspects of Asterisk. We were able to do one type of call transfering, ie, the called person can transfer the original call to another person. but we were unable to do the other, that is, call initiator him/her self couldn't transfer the call. Eventhough the documentation for Dial applicationintructs to use "T" to achieve that. and we learnt that it has not been implemented yet in Asterisk. Is this true? Is some one workin on this issue? if the answer is NO, we can give a try to implement it, with a help of u all , ofcourse :-) (cos, we are quite new to asterisk-only 1 month of experience, but amazed of its great performance) Thank you very much, Surajee
