Re: [asterisk-users] Call pickup on channel sip with SNOM phones issue
On Montag, 27. August 2018 17:42:37 Hans-Peter Jansen wrote: > > What am I missing here, any suggestions? > Okay, scratch it, "notifycid = yes" must reside in the general section! Now, it behaves as expected until: [Aug 27 22:20:37] NOTICE[6200][C-0003]: app_directed_pickup.c:365 pickup_exec: No target channel found for 62@phones Details: extensions.conf: [phones] exten => 60,hint,SIP/60 exten => 61,hint,SIP/61 exten => 62,hint,SIP/62 exten => _60,1,Dial(SIP/60) exten => _61,1,Dial(SIP/61) exten => _62,1,Dial(SIP/62) A call from external to 62 is notified to 60 three times: First a little silly (local and remote are identical): == Extension Changed 62[phones] new state Ringing for Notify User 60 Reliably Transmitting (NAT) to 172.16.23.60:2112: NOTIFY sip:60@172.16.23.60:2112 SIP/2.0 Via: SIP/2.0/UDP 172.16.4.100:5060;branch=z9hG4bK2ec2ef2e;rport Max-Forwards: 70 From: ;tag=as40973611 To: ;tag=ebsb74m178 Contact: Call-ID: 3c95372c15b1-uz42rw4w6sy9 CSeq: 104 NOTIFY User-Agent: Asterisk PBX 15.5.0 Subscription-State: active Event: dialog Content-Type: application/dialog-info+xml Content-Length: 524 sip:62@172.16.4.100 sip:62@172.16.4.100 early <-> Then with an entity of sip:62@172.16.23.8, which is my old asterisk, but with correct local/remote values: --- (7 headers 0 lines) --- Reliably Transmitting (NAT) to 172.16.23.60:2112: NOTIFY sip:60@172.16.23.60:2112 SIP/2.0 Via: SIP/2.0/UDP 172.16.4.100:5060;branch=z9hG4bK60e3a3a9;rport Max-Forwards: 70 From: ;tag=as2c6b3fce To: ;tag=mafy78cezc Contact: Call-ID: 3c95372c1b80-xmqzyr2cq6z2 CSeq: 105 NOTIFY User-Agent: Asterisk PBX 15.5.0 Subscription-State: active Event: dialog Content-Type: application/dialog-info+xml Content-Length: 541 sip:01721234567@172.16.4.100 sip:62@172.16.4.100 early <-> And finally correctly: --- (7 headers 0 lines) --- Reliably Transmitting (NAT) to 172.16.23.60:2112: NOTIFY sip:60@172.16.23.60:2112 SIP/2.0 Via: SIP/2.0/UDP 172.16.4.100:5060;branch=z9hG4bK6d9e9f60;rport Max-Forwards: 70 From: ;tag=as40973611 To: ;tag=ebsb74m178 Contact: Call-ID: 3c95372c15b1-uz42rw4w6sy9 CSeq: 105 NOTIFY User-Agent: Asterisk PBX 15.5.0 Subscription-State: active Event: dialog Content-Type: application/dialog-info+xml Content-Length: 542 sip:01721234567@172.16.4.100 sip:62@172.16.4.100 early <-> NOTIFY Ack: --- (11 headers 0 lines) --- sip_route_dump: route/path hop: -- SIP/62-0003 is ringing <--- SIP read from UDP:172.16.23.60:2112 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 172.16.4.100:5060;branch=z9hG4bK6d9e9f60;rport=5060 From: ;tag=as40973611 To: ;tag=ebsb74m178 Call-ID: 3c95372c15b1-uz42rw4w6sy9 CSeq: 105 NOTIFY Content-Length: 0 <-> 60 want to take over the call: <--- SIP read from UDP:172.16.23.60:2112 ---> INVITE sip:01721234567@172.16.4.100 SIP/2.0 Via: SIP/2.0/UDP 172.16.23.60:2112;branch=z9hG4bK-pb8ywn90vhet;rport From: "HFO" ;tag=omy5lrfdik To: Call-ID: 3c953745720b-p5q7kgcj604q CSeq: 1 INVITE Max-Forwards: 70 Contact: ;reg-id=1 Replaces: pickup-3c95372c15b1-uz42rw4w6sy9;to-tag=as40973611;from-tag=ebsb74m178 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom360/7.3.30 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 452 <-> Unauthorized: --- (19 headers 18 lines) --- Sending to 172.16.23.60:2112 (NAT) [Aug 27 22:20:37] NOTICE[6200][C-0003]: chan_sip.c:26269 handle_request_invite: Trying to pick up 62@phones Sending to 172.16.23.60:2112 (NAT) Using INVITE request as basis request - 3c953745720b-p5q7kgcj604q Found peer '60' for '60' from 172.16.23.60:2112 <--- Reliably Transmitting (NAT) to 172.16.23.60:2112 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.16.23.60:2112;branch=z9hG4bK-pb8ywn90vhet;received=172.16.23.60;rport=2112 From: "HFO" ;tag=omy5lrfdik To: ;tag=as36a783db Call-ID: 3c953745720b-p5q7kgcj604q CSeq: 1 INVITE Server: Asterisk PBX 15.5.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5835824d" Content-Length: 0 <> Ahh, okay <--- SIP read from UDP:172.16.23.60:2112 ---> ACK sip:01721234567@172.16.4.100 SIP/2.0 Via: SIP/2.0/UDP 172.16.23.60:2112;branch=z9hG4bK-pb8ywn90vhet;rport From: "HFO" ;tag=omy5lrfdik To: ;tag=as36a783db Call-ID: 3c953745720b-p5q7kgcj604q CSeq: 1 ACK Max-Forwards: 70 Contact: ;reg-id=1 Content-Length: 0 <-> You want auth, you get auth: <--- SIP read from UDP:172.16.23.60:2112 ---> INVITE sip:01721234567@172.16.4.100 SIP/2.0 Via: SIP/2.0/UDP 172.16.23.60:2112;branch=z9hG4bK-zbjhw9im94ce;rport From: "HFO" ;tag=omy5lrfdik To: Call-ID:
Re: [asterisk-users] Call Pickup how to display CND of incoming number
On Tue, 2013-02-19 at 02:05 +, Klaverstyn, David C wrote: Is it possible to display the incoming calling number on a handset when trying to pick up a call from another handset? I currently have Call Pickup working using *8, I have also used the PickUp application successfully but I’m not sure how to use these features so the handsets show the incoming calling number and not the number that you have dialled to pick up the call. Regards David Klaverstyn Try setting sendrpid to pai in sip.conf -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- 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 Pickup how to display CND of incoming number
- Original Message - From: David C Klaverstyn david.klavers...@intergraph.com Is it possible to display the incoming calling number on a handset when trying to pick up a call from another handset? I currently have Call Pickup working using *8, I have also used the PickUp application successfully but I’m not sure how to use these features so the handsets show the incoming calling number and not the number that you have dialled to pick up the call. You are placing a call *to* Asterisk, therefore the handset, like most will show the number you dialed. I don't know how you would get the CallerID to update during a connected SIP session. I'm no SIP expert, but Googling around - I don't think it's possible, at least easily... http://forums.asterisk.org/viewtopic.php?f=1t=71351p=136777 http://forums.digium.com/viewtopic.php?p=152753 -- Rusty Newton OS Community Support Manager | Digium, Inc. | www.digium.com Office/Cell/Fax: 256-428-6200 -- _ -- 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 Pickup how to display CND of incoming number
Check out connectedline() -Original Message- From: Rusty Newton rnew...@digium.com Sender: asterisk-users-boun...@lists.digium.com Date: Tue, 19 Feb 2013 09:58:30 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Call Pickup how to display CND of incoming number - Original Message - From: David C Klaverstyn david.klavers...@intergraph.com Is it possible to display the incoming calling number on a handset when trying to pick up a call from another handset? I currently have Call Pickup working using *8, I have also used the PickUp application successfully but I’m not sure how to use these features so the handsets show the incoming calling number and not the number that you have dialled to pick up the call. You are placing a call *to* Asterisk, therefore the handset, like most will show the number you dialed. I don't know how you would get the CallerID to update during a connected SIP session. I'm no SIP expert, but Googling around - I don't think it's possible, at least easily... http://forums.asterisk.org/viewtopic.php?f=1t=71351p=136777 http://forums.digium.com/viewtopic.php?p=152753 -- Rusty Newton OS Community Support Manager | Digium, Inc. | www.digium.com Office/Cell/Fax: 256-428-6200 -- _ -- 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 pickup on Asterisk 1.8 and Polycom IP550s?
For what it's worth, the phone is getting enough information. The first call works fine - it's the second call that never triggers the pickup screen, though it does cause the lamp to blink for that line. It's like the phone understands ringing but not busy+ringing. I'm tempted to say it's a Polycom firmware issue, but I haven't seen an errata items that matches. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gord Urquhart Sent: Friday, December 16, 2011 2:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call pickup on Asterisk 1.8 and Polycom IP550s? It sounds like the phone is not getting enough info to do a directed pickup, have you turned on NotifyCID in sip.conf? If that does'nt work try using the extended BLF stuff (described here http://www.excaliburtech.net/archives/147 and here http://www.voip-info.org/wiki/view/Asterisk+presence) gordu On Thu, Dec 15, 2011 at 12:10 PM, Justin Sherrill justin.sherr...@americanrocksalt.commailto:justin.sherr...@americanrocksalt.com wrote: This is one of those Is anyone else doing this?/Is anyone else seeing this? posts. We have an Asterisk 1.8.4 system, with Polycom IP550 phones running firmware 3.2.3. If someone on the 'buddy list' - the list of other extensions to watch - is called, the phone gets a NOTIFY event and displays a screen with the call information and a pickup softkey. However, if someone on that list is already on the phone and they get a second incoming call, the NOTIFY event comes in but the phone never displays the changed screen with the pickup button. It'll flash the light next to that extension, but that's it. Is anyone using a similar setup and seeing this? It's somewhat rare, but I have an office location where everyone there likes to pick up other people's calls, and they haven't been using a call queue like they oughta. Justin Sherrill - American Rock Salt P: 585-991-6825tel:585-991-6825 F: 585-991-6925tel:585-991-6925 C: 585-298-6826tel:585-298-6826 -- _ -- 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 pickup on Asterisk 1.8 and Polycom IP550s?
It sounds like the phone is not getting enough info to do a directed pickup, have you turned on NotifyCID in sip.conf? If that does'nt work try using the extended BLF stuff (described here http://www.excaliburtech.net/archives/147 and here http://www.voip-info.org/wiki/view/Asterisk+presence) gordu On Thu, Dec 15, 2011 at 12:10 PM, Justin Sherrill justin.sherr...@americanrocksalt.com wrote: This is one of those Is anyone else doing this?/Is anyone else seeing this? posts. We have an Asterisk 1.8.4 system, with Polycom IP550 phones running firmware 3.2.3. If someone on the 'buddy list' - the list of other extensions to watch - is called, the phone gets a NOTIFY event and displays a screen with the call information and a pickup softkey. However, if someone on that list is already on the phone and they get a second incoming call, the NOTIFY event comes in but the phone never displays the changed screen with the pickup button. It'll flash the light next to that extension, but that's it. Is anyone using a similar setup and seeing this? It's somewhat rare, but I have an office location where everyone there likes to pick up other people's calls, and they haven't been using a call queue like they oughta. Justin Sherrill - American Rock Salt P: 585-991-6825 F: 585-991-6925 C: 585-298-6826 -- _ -- 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 pickup on Asterisk 1.8 and Polycom IP550s?
AFAIK, Asterisk only picks up the first instance of a line, so if you have 2 calls on exten 100, only the first one is recognized. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Sherrill Sent: Thursday, December 15, 2011 2:11 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Call pickup on Asterisk 1.8 and Polycom IP550s? This is one of those Is anyone else doing this?/Is anyone else seeing this? posts. We have an Asterisk 1.8.4 system, with Polycom IP550 phones running firmware 3.2.3. If someone on the 'buddy list' - the list of other extensions to watch - is called, the phone gets a NOTIFY event and displays a screen with the call information and a pickup softkey. However, if someone on that list is already on the phone and they get a second incoming call, the NOTIFY event comes in but the phone never displays the changed screen with the pickup button. It'll flash the light next to that extension, but that's it. Is anyone using a similar setup and seeing this? It's somewhat rare, but I have an office location where everyone there likes to pick up other people's calls, and they haven't been using a call queue like they oughta. Justin Sherrill - American Rock Salt P: 585-991-6825 F: 585-991-6925 C: 585-298-6826 -- _ -- 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 pickup
On 10/05/2011 11:21 PM, A. M. Hoffmeister wrote: Am 05.10.2011 20:42, schrieb Marek Cervenka: hello, is there some way to notify people in the same pickup group about call from caller to callee? i.e. i have call from 111 to 222 there are 222,333,444 in the same pickup group 333,444 see on the phone (aastra) that 111 calling to 222 and can pickup the call with *8 siemens have this on their sip openstage phones. how they do this? You can have that with subscriptions/hints, for example Snom phones can display not only a call to one of the peers but also the caller and callee identification. can you point me to some doc/examples? how this is implemented in SIP? i think about sending some notify from dialplan (i have incoming call, i know who is in pickup group, i can send call to callee and before send some NOTIFY to other phones in the pickupgroup) i found only one app like this - jabbersend. but i need this notification on phone screen This works jaw to cheek with BLF (busy lamp field) which allows to monitor other extensions' status (in_use, ringing...). Of course you can be member of a pickup group without monitoring the status of any of the peers, and you can monitor a peer's status without being in the same pickup group (although not pickup the call then, obviously :-) -- --- Marek Cervenka Centrum Vypocetni Techniky jabber - cerv...@njs.netlab.cz CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz RHCE 100-175-678 === -- _ -- 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 pickup
Search for dialog-info pickup -Original Message- From: Marek Cervenka cerv...@fpf.slu.cz Sender: asterisk-users-boun...@lists.digium.com Date: Fri, 07 Oct 2011 09:47:45 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] call pickup On 10/05/2011 11:21 PM, A. M. Hoffmeister wrote: Am 05.10.2011 20:42, schrieb Marek Cervenka: hello, is there some way to notify people in the same pickup group about call from caller to callee? i.e. i have call from 111 to 222 there are 222,333,444 in the same pickup group 333,444 see on the phone (aastra) that 111 calling to 222 and can pickup the call with *8 siemens have this on their sip openstage phones. how they do this? You can have that with subscriptions/hints, for example Snom phones can display not only a call to one of the peers but also the caller and callee identification. can you point me to some doc/examples? how this is implemented in SIP? i think about sending some notify from dialplan (i have incoming call, i know who is in pickup group, i can send call to callee and before send some NOTIFY to other phones in the pickupgroup) i found only one app like this - jabbersend. but i need this notification on phone screen This works jaw to cheek with BLF (busy lamp field) which allows to monitor other extensions' status (in_use, ringing...). Of course you can be member of a pickup group without monitoring the status of any of the peers, and you can monitor a peer's status without being in the same pickup group (although not pickup the call then, obviously :-) -- --- Marek Cervenka Centrum Vypocetni Techniky jabber - cerv...@njs.netlab.cz CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz RHCE 100-175-678 === -- _ -- 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 Pickup with 1.6.2.1 and Snom
Hi, I've experienced the same thing in the 1.6.2 release, with the 1.6.1 all work as expected. There is nothing in the changelog ... So, I think it's a bug ? -Message d'origine- De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de Loris Santamaria Envoyé : samedi 13 février 2010 04:09 À : asterisk-users@lists.digium.com Objet : [asterisk-users] Call Pickup with 1.6.2.1 and Snom Hi, I've used various patches with asterisk 1.4 to have support for call pickup and notification with good results. Now I'm trying vanilla 1.6.2 with its official support for dialog-info +xml notifications with no success. This is what i'm doing: - Phone A has a key configured as type extension pointing to Phone B. - In sip.conf I added notifycid=ignore-context - Phone A and B and C are in the same callgroup and pickupgroup - Phone A and B and C are in the same context Phone C calls Phone B and asterisk generates a notification for phone A: ?xml version=1.0? dialog-info xmlns=urn:ietf:params:xml:ns:dialog-info version=11 state=full entity=sip:35...@10.40.23.179 dialog id=35505 call-id=pickup-3c26701519b8-5xxapzoav2u4 direction=recipient remote identity display=Lab 1sip:35...@10.40.23.179/identity target uri=sip:35...@10.40.23.179/ /remote local identitysip:35...@10.40.23.179/identity target uri=sip:35...@10.40.23.179/ /local stateearly/state /dialog /dialog-info With this notification, Phone A shows on the screen that Phone C is calling Phone B, and the function key blinks. If one presses the blinking function key, the phone generates an Invite with replaces, to try to pickup the call: INVITE sip:35...@10.40.23.179 SIP/2.0 Via: SIP/2.0/UDP 10.40.24.175:5060;branch=z9hG4bK-qoz3zjhmyfcw;rport From: Lab 4 sip:35...@10.40.23.179;tag=o28fq65rfu To: Lab 1 sip:35...@10.40.23.179 Call-ID: 3c2672b3f35a-dpd0zv11yegl CSeq: 1 INVITE Max-Forwards: 70 Contact: sip:35...@10.40.24.175:5060;flow-id=1 Replaces: pickup-3c26701519b8-5xxapzoav2u4 P-Key-Flags: keys=3 User-Agent: snom320/7.1.39 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 368 Then asterisk receives the pickup request: [Feb 11 10:44:13] DEBUG[4659] chan_sip.c: Invite/replaces: Will use Replace-Call-ID : pickup-3c26701519b8-5xxapzoav2u4 Fromtag: no from tag Totag: no to tag [Feb 11 10:44:13] NOTICE[4659] chan_sip.c: Trying to pick up 35...@rededelca [Feb 11 10:44:13] VERBOSE[4659] chan_sip.c: Sending to 10.40.24.175 : 5060 (no NAT) [Feb 11 10:44:13] DEBUG[4659] chan_sip.c: Initializing initreq for method INVITE - callid 3c2672b3f35a-dpd0zv11yegl [Feb 11 10:44:13] VERBOSE[4659] chan_sip.c: Using INVITE request as basis request - 3c2672b3f35a-dpd0zv11yegl [...] [Feb 11 10:44:13] DEBUG[4659] chan_sip.c: INVITE part of call transfer. Replaces [pickup-3c26701519b8-5xxapzoav2u4] [Feb 11 10:44:13] DEBUG[4659] chan_sip.c: Invite/replaces: Will use Replace-Call-ID : pickup-3c26701519b8-5xxapzoav2u4 Fromtag: no from tag Totag: no to tag [Feb 11 10:44:13] NOTICE[4659] chan_sip.c: Trying to pick up 35...@rededelca [Feb 11 10:44:13] VERBOSE[4659] chan_sip.c: Sending to 10.40.24.175 : 5060 (no NAT) [Feb 11 10:44:13] DEBUG[4659] chan_sip.c: Initializing initreq for method INVITE - callid 3c2672b3f35a-dpd0zv11yegl [Feb 11 10:44:13] VERBOSE[4659] chan_sip.c: Using INVITE request as basis request - 3c2672b3f35a-dpd0zv11yegl [...] [Feb 11 10:44:13] DEBUG[4659] chan_sip.c: About to call Pickup(35...@pickupmark) [Feb 11 10:44:13] DEBUG[4649] devicestate.c: Changing state for SIP/35504 - state 2 (In use) [Feb 11 10:44:13] DEBUG[4649] devicestate.c: device 'SIP/35504' state '2' [Feb 11 10:44:13] NOTICE[4659] app_directed_pickup.c: No target channel found for 35505. [Feb 11 10:44:13] DEBUG[4659] channel.c: Hanging up channel 'SIP/35504-000f' [Feb 11 10:44:13] DEBUG[4659] chan_sip.c: Hangup call SIP/35504-000f, SIP callid 3c2672b3f35a-dpd0zv11yegl After this obviously phone A hasn't picked up the call, and Phone B keeps ringing. Did I miss something in the dialplan or is it a bug? -- Loris Santamaria linux user #70506 xmpp:lo...@lgs.com.ve Links Global Services, C.A.http://www.lgs.com.ve Tel: 0286 952.06.87 Cel: 0414 095.00.10 sip:1...@lgs.com.ve -O9 -omg-optimize -fomit-instructions -- _ -- 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 --
Re: [asterisk-users] Call Pickup with 1.6.2.1 and Snom
hi friend need ur help in dial plan, i want to allow exten 2000 to 2005 can
make call outside and exten 2006 to 2010 can not make call outside. heres my
dial plan.
sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0
context = others
[2000]
type=friend
context=outside
secret=1234
host=dynamic
[2001]
type=friend
context=outside
secret=1234
host=dynamic
[2002]
type=friend
context=outside
secret=1234
host=dynamic
[2003]
type=friend
contex=outside
secret=1234
host=dynamic
[2004]
type=friend
contex=outside
secret=1234
host=dynamic
[2005]
type=friend
contex=outside
secret=1234
host=dynamic
[2006]
type=friend
contex=internal
secret=1234
host=dynamic
[2007]
type=friend
contex=internal
secret=1234
host=dynamic
[2008]
type=friend
contex=internal
secret=1234
host=dynamic
[2009]
type=friend
contex=internal
secret=1234
host=dynamic
[2010]
type=friend
contex=internal
secret=1234
host=dynamic
vi /etc/asterisk/extensions.conf
[from-zaptel]
exten = s,1,wait(2)
exten = s,n,dial(sip/2000)
exten = s,n,dial(sip/2001)
exten = s,n,Playback(tt-weasels)
[others]
include = internal
include = outside
[inside]
exten = _20XX,1,Dial(SIP/${EXTEN})
exten = _20XX,n,VoiceMail(${ext...@others,u)
exten = _20XX,n,Hangup()
[outside]
exten = 2001,1,Dial(Zap/1-1/${EXTEN})
exten = 2001,n,Hangup
exten = 2002,1,Dial(Zap/1-1/${EXTEN})
exten = 2002,n,Hangup
exten = 2003,1,Dial(Zap/1-1/${EXTEN})
exten = 2003,n,Hangup
exten = 2004,1,Dial(Zap/1-1/${EXTEN})
exten = 2004,n,Hangup
exten = 2005,1,Dial(Zap/1-1/${EXTEN})
exten = 2005,n,Hangup
this is the log when i am calling from exten 2000 to outside
Connected to Asterisk 1.4.29 currently running on localhost (pid = 2243)
Verbosity is at least 3
[Feb 13 12:05:47] NOTICE[2482]: chan_sip.c:15124 handle_request_invite: Call
from '2002' to extension '919369613616' rejected because extension not found.
any help n support will be highly appreciated
--- On Sat, 13/2/10, Loris Santamaria lo...@lgs.com.ve wrote:
From: Loris Santamaria lo...@lgs.com.ve
Subject: [asterisk-users] Call Pickup with 1.6.2.1 and Snom
To: asterisk-users@lists.digium.com
Date: Saturday, 13 February, 2010, 8:39 AM
Hi,
I've used various patches with asterisk 1.4 to have support for call
pickup and notification with good results.
Now I'm trying vanilla 1.6.2 with its official support for dialog-info
+xml notifications with no success. This is what i'm doing:
- Phone A has a key configured as type extension pointing to Phone B.
- In sip.conf I added notifycid=ignore-context
- Phone A and B and C are in the same callgroup and pickupgroup
- Phone A and B and C are in the same context
Phone C calls Phone B and asterisk generates a notification for phone A:
?xml version=1.0?
dialog-info xmlns=urn:ietf:params:xml:ns:dialog-info version=11
state=full entity=sip:35...@10.40.23.179
dialog id=35505 call-id=pickup-3c26701519b8-5xxapzoav2u4
direction=recipient
remote
identity display=Lab 1sip:35...@10.40.23.179/identity
target uri=sip:35...@10.40.23.179/
/remote
local
identitysip:35...@10.40.23.179/identity
target uri=sip:35...@10.40.23.179/
/local
stateearly/state
/dialog
/dialog-info
With this notification, Phone A shows on the screen that Phone C is
calling Phone B, and the function key blinks. If one presses the
blinking function key, the phone generates an Invite with replaces, to
try to pickup the call:
INVITE sip:35...@10.40.23.179 SIP/2.0
Via: SIP/2.0/UDP 10.40.24.175:5060;branch=z9hG4bK-qoz3zjhmyfcw;rport
From: Lab 4 sip:35...@10.40.23.179;tag=o28fq65rfu
To: Lab 1 sip:35...@10.40.23.179
Call-ID: 3c2672b3f35a-dpd0zv11yegl
CSeq: 1 INVITE
Max-Forwards: 70
Contact: sip:35...@10.40.24.175:5060;flow-id=1
Replaces: pickup-3c26701519b8-5xxapzoav2u4
P-Key-Flags: keys=3
User-Agent: snom320/7.1.39
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 368
Then asterisk receives the pickup request:
[Feb 11 10:44:13] DEBUG[4659] chan_sip.c: Invite/replaces: Will use
Replace-Call-ID : pickup-3c26701519b8-5xxapzoav2u4 Fromtag: no from
tag Totag: no to tag
[Feb 11 10:44:13] NOTICE[4659] chan_sip.c: Trying to pick up 35...@rededelca
[Feb 11 10:44:13] VERBOSE[4659] chan_sip.c: Sending to 10.40.24.175 : 5060 (no
NAT)
[Feb 11 10:44:13] DEBUG[4659] chan_sip.c: Initializing initreq for method
INVITE - callid 3c2672b3f35a-dpd0zv11yegl
[Feb 11 10:44:13] VERBOSE[4659] chan_sip.c: Using INVITE request as basis
request - 3c2672b3f35a-dpd0zv11yegl
[...]
[Feb 11 10:44:13] DEBUG[4659] chan_sip.c: INVITE part of call transfer.
Replaces [pickup-3c26701519b8-5xxapzoav2u4]
[Feb 11 10:44:13] DEBUG[4659] chan_sip.c: Invite/replaces: Will use
Replace-Call-ID : pickup-3c26701519b8-5xxapzoav2u4 Fromtag: no from
Re: [asterisk-users] Call Pickup Works w/Linksys ATA, not with Cisco 7940G
On Tue, 7 Apr 2009, George Pajari wrote: I have an Asterisk 1.4.18 with a mix of cordless phones connected using Linksys SPA2102 ATAs and Cisco 7940G phones. Unit obtains SIP trunking from an ITSP (server has no PCI boards). *8 Call Pickup works fine from any of the phones connected using the Linksys SPA2102. *8 Call Pickup does not work from the Cisco 7940G phones (chan_sip.c:13977 handle_request_invite: Nothing to pick up for 000d6556-eeb3001c-76b88543-7f51d...@192.168.0.211) Seems someone else had the same problem back in 2004 and got no answer. http://lists.digium.com/pipermail/asterisk-users/2004-April/036869.html Vincent Li System Administrator BRC,UBC perl -e'print\131e\164\040\101n\157t\150e\162\040\114i\156u\170\040\107e\145k\012' ___ -- 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 pickup and ring groups
--- On Fri, 3/6/09, Vieri rentor...@yahoo.com wrote: I'm having trouble with call pickups. Suppose ring group is 100 and has extensions 101 and 102. Someone calls 100, 101 rings and 102 wants to pick the call up. If 102 dials **100, call pickup works. If 102 dials **101, call pickup fails. In my dialplan I have: exten = **101,1,NoOp(pickup extension) exten = **101,n,Pickup(101) exten = **101,n,NoOp(pickup group) exten = **101,n,Pickup(100) exten = **101,n,Hangup When 102 dials **101 I see this on the CLI: -- SIP/4060-08868de8 is ringing Extension Changed 4060 new state Ringing for Notify User 4061 -- Executing NoOp(SIP/4053-0886ba08, pickup extension) in new stack -- Executing Pickup(SIP/4053-0886ba08, 4060) in new stack == Spawn extension (from-internal, **4060, 2) exited non-zero on 'SIP/4053-0886ba08' It does NOT continue and display pickup group so it just hangs up the call. It *should* go on and reach the Pickup(100) instruction... Why is it failing? I've noticed this only after I recently upgraded from Asterisk 1.2.30 to 1.2.31.1. Asterisk 1.4.21.2 does not have this bug. Can someone please let me know if the 1.2 branch can be fixed (should I file a bug report or will it be ignored since 1.2 only has security fixes)? Thanks Sorry for the CLI mix-up: in my original example, 4053 is extension 102 and 4060 is 101. ___ -- 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 pickup - Asterisk 1.4.19.1 -
works very well , features.conf 2008/5/1 Jose P. Espinal [EMAIL PROTECTED]: Hello List, Does anyone here have call pickup (with *8 ) working ok on Asterisk version 1.4.19.1 ? Thanks in advice, -- ___ -- 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 pickup - Asterisk 1.4.19.1 -
Call pickup (defaults to *8) does not work for IAX2 channels. troxlinux wrote: works very well , features.conf 2008/5/1 Jose P. Espinal [EMAIL PROTECTED]: Hello List, Does anyone here have call pickup (with *8 ) working ok on Asterisk version 1.4.19.1 ? -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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 Pickup with more than one argument
On Wed, Apr 11, 2007 at 05:27:51PM +0100, Ricardo Carvalho wrote: Dear all, Does Pickup application accept multiple extensions pickup syntax, like the following line? Pickup(extension1extension2...) I've tried it in Asterisk 1.2.17 but it doesn't work. Does it work in Asterisk 1.4 already? Or is any other way in any version of Asterisk that I can use to do the same thing? I believe that the Bristuff ChanPickup supports this. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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-pickup function in Queue application
Hi Attilla, I'm not sure if there is something like that available or not, but I know there are some alternatives. You can set the time out limit to say 15 seconds, which for me is about 3-4 rings on the phone before it goes looking for the next agent. The other option you can manually remove the interface from the queue via the CLI by the following: remove queue member Interface from queue name However, I'm not sure if that will have an effect on the call...hopefully it will just send the caller looking for the next number. I haven't personally tried it. I know some phones like the Polycom 601 have a buddy watch option. As far as I know, and someone can step in and correct me if I am wrong, that will just show if the person is on the phone or not. I don't think you can pick up on the line. Kevin Attilla De Groot wrote: Hi All, I need a function that I believe isn't available in Asterisk, but I don't know if I'm correct about this. I have a queue and I want agents that are in that queue to have the ability to answer a call in the queue with calling an extention. For example, if I'm an agent and my colleague forgot to logout I could take the call when his phone is still ringing without walking to his desk or waiting for round robin. Can anyone tell me if this already is avalible ? Regards, Attilla ___ --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-pickup function in Queue application
On Jun 4, 2006, at 10:33 PM, Kevin Smith wrote: Hi Attilla, I'm not sure if there is something like that available or not, but I know there are some alternatives. You can set the time out limit to say 15 seconds, which for me is about 3-4 rings on the phone before it goes looking for the next agent. The other option you can manually remove the interface from the queue via the CLI by the following: remove queue member Interface from queue name However, I'm not sure if that will have an effect on the call...hopefully it will just send the caller looking for the next number. I haven't personally tried it. I know some phones like the Polycom 601 have a buddy watch option. As far as I know, and someone can step in and correct me if I am wrong, that will just show if the person is on the phone or not. I don't think you can pick up on the line. Kevin Hi Kevin, Well I thought about those alternatives and I suggested them, but the person who wants them said that such a feature was avalible on another pbx where he used to work. And well, he would like the same thing on the Asterisk PBX. I already have the time at 15 seconds, and well removing a member from the queue might send it to the next agent. But if there are more then two agents in the queue there is not really a point. Regards, Attilla ___ --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-pickup function in Queue application
Attilla De Groot wrote: Hi All, I have a queue and I want agents that are in that queue to have the ability to answer a call in the queue with calling an extention. For example, if I'm an agent and my colleague forgot to logout I could take the call when his phone is still ringing without walking to his desk or waiting for round robin. http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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-pickup function in Queue application
Attilla De Groot wrote: On Jun 4, 2006, at 10:33 PM, Kevin Smith wrote: Hi Attilla, I'm not sure if there is something like that available or not, but I know there are some alternatives. You can set the time out limit to say 15 seconds, which for me is about 3-4 rings on the phone before it goes looking for the next agent. The other option you can manually remove the interface from the queue via the CLI by the following: remove queue member Interface from queue name However, I'm not sure if that will have an effect on the call...hopefully it will just send the caller looking for the next number. I haven't personally tried it. I know some phones like the Polycom 601 have a buddy watch option. As far as I know, and someone can step in and correct me if I am wrong, that will just show if the person is on the phone or not. I don't think you can pick up on the line. Kevin Hi Kevin, Well I thought about those alternatives and I suggested them, but the person who wants them said that such a feature was avalible on another pbx where he used to work. And well, he would like the same thing on the Asterisk PBX. I already have the time at 15 seconds, and well removing a member from the queue might send it to the next agent. But if there are more then two agents in the queue there is not really a point. Depending on the device type could you not use call pickups with *8? Not sure if it works with queues, but it definitely works with normal calls. http://www.voip-info.org/wiki-PBX+Call+Pickup -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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 Pickup Woes
What about setting up DYNAMIC_FEATURES=pickupexten inside your [globals] ? This is needed for, as the variable name says, dynamic features. And don't forget to set callgroup/pickupgroup to each one in your sip.conf Does anyone tested the new application Pickup()? []'s MM On Mon, 2006-03-20 at 09:24 +1100, Adam Dale wrote: Hello all, I have an asterisk @ home system running 1.2.4. Call pickup seems to be a bit of a problem. I’ve looked at a lot of posts and the wiki, which states that you need to define the pickup extension in features.conf and the pickup groups in sip.conf. I’ve done this, however there is no definition for *8 in extensions.conf. Is there supposed to be and it has been removed? ___ --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 Pickup Woes
Melcon Moraes wrote: On Mon, 2006-03-20 at 09:24 +1100, Adam Dale wrote: Hello all, I have an asterisk @ home system running 1.2.4. Call pickup seems to be a bit of a problem. I’ve looked at a lot of posts and the wiki, which states that you need to define the pickup extension in features.conf and the pickup groups in sip.conf. I’ve done this, however there is no definition for *8 in extensions.conf. I've confirmed this morning. Call pickup is broken in 1.24. I've upgraded our system to 1.25 over the weekend and tested out call pickup this morning. It now works. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call Pickup Woes
And don't forget to set callgroup/pickupgroup to each one in your sip.conf Call pickup works among IAX phones? Mimmus ___ --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 Pickup Woes
There shouldn't be one, have you tried it? what is the CLI output? On 3/19/06, Adam Dale [EMAIL PROTECTED] wrote: Hello all, I have an asterisk @ home system running 1.2.4. Call pickup seems to be a bit of a problem. I've looked at a lot of posts and the wiki, which states that you need to define the pickup extension in features.conf and the pickup groups in sip.conf. I've done this, however there is no definition for *8 in extensions.conf. Is there supposed to be and it has been removed? ___ --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 Pickup Woes
C F wrote: groups in sip.conf. I've done this, however there is no definition for *8 in extensions.conf. Its not in extensions.conf, its in features.conf -- in extensions.conf you have to configure callgroups for each of your extensions, so that you can pick them up with *8. -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.net ./ ___ --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 Pickup Woes
I've configured the following in features.conf pickupexten = *8 ; Configure the pickup extension. Default is *8 and all SIP extensions are configured as pickupgroup=1. These phones can make and receive calls, and also use features such as *69, *70 and *98. When I dial *8 I get a beeping as if there is no valid extension and no debugging information when I open the console with asterisk -vvvr -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Avi Miller Sent: Monday, 20 March 2006 9:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Pickup Woes C F wrote: groups in sip.conf. I've done this, however there is no definition for *8 in extensions.conf. Its not in extensions.conf, its in features.conf -- in extensions.conf you have to configure callgroups for each of your extensions, so that you can pick them up with *8. -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.net ./ ___ --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 Pickup Woes
You have to configre the Dialplan in your sip phone to accept *8 What phone are you using? On 3/19/06, Adam Dale [EMAIL PROTECTED] wrote: I've configured the following in features.conf pickupexten = *8 ; Configure the pickup extension. Default is *8 and all SIP extensions are configured as pickupgroup=1. These phones can make and receive calls, and also use features such as *69, *70 and *98. When I dial *8 I get a beeping as if there is no valid extension and no debugging information when I open the console with asterisk -vvvr -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Avi Miller Sent: Monday, 20 March 2006 9:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Pickup Woes C F wrote: groups in sip.conf. I've done this, however there is no definition for *8 in extensions.conf. Its not in extensions.conf, its in features.conf -- in extensions.conf you have to configure callgroups for each of your extensions, so that you can pick them up with *8. -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.net ./ ___ --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 Pickup Woes
I am using Cisco 7940/60/70's -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Monday, 20 March 2006 10:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Pickup Woes You have to configre the Dialplan in your sip phone to accept *8 What phone are you using? On 3/19/06, Adam Dale [EMAIL PROTECTED] wrote: I've configured the following in features.conf pickupexten = *8 ; Configure the pickup extension. Default is *8 and all SIP extensions are configured as pickupgroup=1. These phones can make and receive calls, and also use features such as *69, *70 and *98. When I dial *8 I get a beeping as if there is no valid extension and no debugging information when I open the console with asterisk -vvvr -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Avi Miller Sent: Monday, 20 March 2006 9:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Pickup Woes C F wrote: groups in sip.conf. I've done this, however there is no definition for *8 in extensions.conf. Its not in extensions.conf, its in features.conf -- in extensions.conf you have to configure callgroups for each of your extensions, so that you can pick them up with *8. -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.net ./ ___ --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 Pickup Woes
Now I'm sure it's a dialplan problem, configure your dialplan to allow *8. You can do that in the SIPDefault.cnf file On 3/19/06, Adam Dale [EMAIL PROTECTED] wrote: I am using Cisco 7940/60/70's -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Monday, 20 March 2006 10:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Pickup Woes You have to configre the Dialplan in your sip phone to accept *8 What phone are you using? On 3/19/06, Adam Dale [EMAIL PROTECTED] wrote: I've configured the following in features.conf pickupexten = *8 ; Configure the pickup extension. Default is *8 and all SIP extensions are configured as pickupgroup=1. These phones can make and receive calls, and also use features such as *69, *70 and *98. When I dial *8 I get a beeping as if there is no valid extension and no debugging information when I open the console with asterisk -vvvr -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Avi Miller Sent: Monday, 20 March 2006 9:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Pickup Woes C F wrote: groups in sip.conf. I've done this, however there is no definition for *8 in extensions.conf. Its not in extensions.conf, its in features.conf -- in extensions.conf you have to configure callgroups for each of your extensions, so that you can pick them up with *8. -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.net ./ ___ --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 ___ --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 Pickup Woes
C F wrote: Now I'm sure it's a dialplan problem, configure your dialplan to allow *8. You can do that in the SIPDefault.cnf file On 3/19/06, Adam Dale [EMAIL PROTECTED] wrote: I am using Cisco 7940/60/70's Don't you mean the dialplan.xml. This is what I have: DIALTEMPLATE TEMPLATE MATCH=*Timeout=5/ !-- Anything else -- TEMPLATE MATCH=#Timeout=5/ !-- Anything else -- /DIALTEMPLATE -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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 Pickup Woes
Thank you very much. I'll now investigate how to set up dialplan.xml. I've never had to set it up before. Cheers, Much appreciated. :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Monday, 20 March 2006 11:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Pickup Woes Now I'm sure it's a dialplan problem, configure your dialplan to allow *8. You can do that in the SIPDefault.cnf file On 3/19/06, Adam Dale [EMAIL PROTECTED] wrote: I am using Cisco 7940/60/70's -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Monday, 20 March 2006 10:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Pickup Woes You have to configre the Dialplan in your sip phone to accept *8 What phone are you using? On 3/19/06, Adam Dale [EMAIL PROTECTED] wrote: I've configured the following in features.conf pickupexten = *8 ; Configure the pickup extension. Default is *8 and all SIP extensions are configured as pickupgroup=1. These phones can make and receive calls, and also use features such as *69, *70 and *98. When I dial *8 I get a beeping as if there is no valid extension and no debugging information when I open the console with asterisk -vvvr -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Avi Miller Sent: Monday, 20 March 2006 9:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Pickup Woes C F wrote: groups in sip.conf. I've done this, however there is no definition for *8 in extensions.conf. Its not in extensions.conf, its in features.conf -- in extensions.conf you have to configure callgroups for each of your extensions, so that you can pick them up with *8. -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.net ./ ___ --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 ___ --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 Pickup Woes
in AAH you can set the callgroup and pickup group within each extensions setup. On 3/19/06, Adam Dale [EMAIL PROTECTED] wrote: Thank you very much. I'll now investigate how to set up dialplan.xml. I've never had to set it up before. Cheers, Much appreciated. :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Monday, 20 March 2006 11:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Pickup Woes Now I'm sure it's a dialplan problem, configure your dialplan to allow *8. You can do that in the SIPDefault.cnf file On 3/19/06, Adam Dale [EMAIL PROTECTED] wrote: I am using Cisco 7940/60/70's -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Monday, 20 March 2006 10:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Pickup Woes You have to configre the Dialplan in your sip phone to accept *8 What phone are you using? On 3/19/06, Adam Dale [EMAIL PROTECTED] wrote: I've configured the following in features.conf pickupexten = *8 ; Configure the pickup extension. Default is *8 and all SIP extensions are configured as pickupgroup=1. These phones can make and receive calls, and also use features such as *69, *70 and *98. When I dial *8 I get a beeping as if there is no valid extension and no debugging information when I open the console with asterisk -vvvr -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Avi Miller Sent: Monday, 20 March 2006 9:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Pickup Woes C F wrote: groups in sip.conf. I've done this, however there is no definition for *8 in extensions.conf. Its not in extensions.conf, its in features.conf -- in extensions.conf you have to configure callgroups for each of your extensions, so that you can pick them up with *8. -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.net ./ ___ --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 ___ --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 -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --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 Pickup Woes
H, I'm still a little stumped. I edited SIPDefault to and created a dialplan.xml file which is being uploaded to the phone. Still no output on the asterisk console wheh I dial *8. :( dialplan.xml DIALTEMPLATE TEMPLATE MATCH=*Timeout=5/ !-- Anything else -- /DIALTEMPLATE SIPDefault.cnf extract: # XML file that specifies the dialplan desired dial_template: dialplan :( -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Monday, 20 March 2006 12:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Pickup Woes C F wrote: Now I'm sure it's a dialplan problem, configure your dialplan to allow *8. You can do that in the SIPDefault.cnf file On 3/19/06, Adam Dale [EMAIL PROTECTED] wrote: I am using Cisco 7940/60/70's Don't you mean the dialplan.xml. This is what I have: DIALTEMPLATE TEMPLATE MATCH=*Timeout=5/ !-- Anything else -- TEMPLATE MATCH=#Timeout=5/ !-- Anything else -- /DIALTEMPLATE -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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 Pickup Woes
You don't need to mess with the dialplan.xml on a cisco phone. Try dialing *8# to pick up a ringing phone. It works just fine here with nothing special in features.conf or extensions.conf. Adam Dale wrote: H, I'm still a little stumped. I edited SIPDefault to and created a dialplan.xml file which is being uploaded to the phone. Still no output on the asterisk console wheh I dial *8. :( dialplan.xml DIALTEMPLATE TEMPLATE MATCH=*Timeout=5/ !-- Anything else -- /DIALTEMPLATE SIPDefault.cnf extract: # XML file that specifies the dialplan desired dial_template: dialplan :( -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Monday, 20 March 2006 12:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Pickup Woes C F wrote: Now I'm sure it's a dialplan problem, configure your dialplan to allow *8. You can do that in the SIPDefault.cnf file On 3/19/06, Adam Dale [EMAIL PROTECTED] wrote: I am using Cisco 7940/60/70's Don't you mean the dialplan.xml. This is what I have: DIALTEMPLATE TEMPLATE MATCH=*Timeout=5/ !-- Anything else -- TEMPLATE MATCH=#Timeout=5/ !-- Anything else -- /DIALTEMPLATE ___ --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 Pickup Woes
Unfortunatly I get a beeping sound and that's it. Just like when I dial something that does not have a match in extensions.conf :( -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Monday, 20 March 2006 1:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Pickup Woes You don't need to mess with the dialplan.xml on a cisco phone. Try dialing *8# to pick up a ringing phone. It works just fine here with nothing special in features.conf or extensions.conf. Adam Dale wrote: H, I'm still a little stumped. I edited SIPDefault to and created a dialplan.xml file which is being uploaded to the phone. Still no output on the asterisk console wheh I dial *8. :( dialplan.xml DIALTEMPLATE TEMPLATE MATCH=*Timeout=5/ !-- Anything else -- /DIALTEMPLATE SIPDefault.cnf extract: # XML file that specifies the dialplan desired dial_template: dialplan :( -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Monday, 20 March 2006 12:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Pickup Woes C F wrote: Now I'm sure it's a dialplan problem, configure your dialplan to allow *8. You can do that in the SIPDefault.cnf file On 3/19/06, Adam Dale [EMAIL PROTECTED] wrote: I am using Cisco 7940/60/70's Don't you mean the dialplan.xml. This is what I have: DIALTEMPLATE TEMPLATE MATCH=*Timeout=5/ !-- Anything else -- TEMPLATE MATCH=#Timeout=5/ !-- Anything else -- /DIALTEMPLATE ___ --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 Pickup with Dialog on snom display
hello bastian, you could use the patch i made http://bugs.digium.com/view.php?id=5014 frank Bastian Schern schrieb: I'm using the snom Phones together with Asterisk and I already able to see which Peer is used via hint priority. Then a LED on the snom phone is blinking. But I don't see who is calling the other phone. I know that the snom phones are already support this feature. But how I can enable this on Asterisk? ___ --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 pickup with Sipura-3000
When I pick up calls on my Sipura I just dial *8# instead of *8. The # will end the Sipura's dial plan. If you put *8 into the dialplan, that would work too. --On Saturday, February 26, 2005 11:39 PM -0700 Joseph [EMAIL PROTECTED] wrote: On Sat, 2005-02-26 at 18:22 -0700, Joseph wrote: I can not make a call pickup to work with Sipura-3000. I have one SIP phone and one is connected to ATA Sipura-3000 I've in all sip.conf context callgroup=1 pickupgroup=1 in features.conf I've tired: pickupexten = *88 pickupexten = *8 Nothing works. What am I missing? I found it! It can be solved by defining: pickupexten = 33 ;any unique number or in Line 1 dia plan (xx.|*xx) ;this permits passing *8 through Line1 -- # Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call pickup with Sipura-3000
On Sat, 2005-02-26 at 18:22 -0700, Joseph wrote: I can not make a call pickup to work with Sipura-3000. I have one SIP phone and one is connected to ATA Sipura-3000 I've in all sip.conf context callgroup=1 pickupgroup=1 in features.conf I've tired: pickupexten = *88 pickupexten = *8 Nothing works. What am I missing? I found it! It can be solved by defining: pickupexten = 33 ;any unique number or in Line 1 dia plan (xx.|*xx) ;this permits passing *8 through Line1 Or, without the dial plan change, just dial *8# like the wiki suggests. The # in this case says I'm done dialing, now send the digits to asterisk. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call pickup with Sipura-3000
On Sat, 2005-02-26 at 18:22 -0700, Joseph wrote: I can not make a call pickup to work with Sipura-3000. I have one SIP phone and one is connected to ATA Sipura-3000 I've in all sip.conf context callgroup=1 pickupgroup=1 in features.conf I've tired: pickupexten = *88 pickupexten = *8 Nothing works. What am I missing? I found it! It can be solved by defining: pickupexten = 33 ;any unique number or in Line 1 dia plan (xx.|*xx) ;this permits passing *8 through Line1 -- #Joseph ___ 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 pickup across technologies (SIP, IAX, MGCP)?
Hi again! it appears that call pick-up only works _within_ a technolgoy, i.e. with a SIP phone when another SIP phone is ringing. Is that correct, or is my configuration faulty? * Case 2: IAX phone ringing - SIP phone can't pick the call up: NOTICE[10250]: Nothing to pick up This seems less a matter of technology than a matter of implementation. From the SIP phones, I can pickup ANY call, no matter if between ISDN, SIP or cross-channel. From the ISDN phones, I can pickup NO calls (unknown extension *8 in context from_ISDN). Hm... with the help of the bristuff PickUp() app I was able to solve this unkown extension for 2 of my 3 cases, but trying to pickup a ringing IAX phone with SIP still fails with error no channel found 2 (bristuff exten = *8,1,PickUp(1)). All clients have callgroup=1 and pickupgroup=1. If I do ship show peer peername I get: Callgroup: 1 (2) Pickupgroup : 1 (2) and I wonder what the (2) is supposed to mean in both cases, the errormessage as well as the peer info. Maybe there is a difference in implementation of callgroup= in iax.conf where one starts couting at 0 and the other at 1? Hm... too bad there is no iax2 show peer peername... Philipp ___ 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 Pickup
You also need callgroup=0 in the sip.conf per user as well. callgroup = the group this sip entry belongs to pickupgroup = the group(s) this sip entry is allowed to pickup Julian. Roger Schreiter wrote: Hi, I put a pickupgroup=0 line for each user in sip.conf. After restarting asterisk I called my collegues phone with my cell phone, I heard it ringing and saw ringing in the asterisk console. Then I dialed *8 with my phone and got on the console: Jan 24 20:41:45 NOTICE[13747]: chan_sip.c:7321 handle_request: Nothing to pick up -- SIP/collegue-92e5 is ringing while the other phone kept ringing. I'm using asterisk-1.0.3 What went wrong? Thanks for any hints! Roger. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Pickup
Roger Schreiter wrote: Hi, I put a pickupgroup=0 line for each user in sip.conf. As far as I'm aware, pickup groups are only for zap interfaces... -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ 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 Pickup
Matt, they work fine on zap and sip. I wish they worked on IAX. Ernie On Jan 24, 2005, at 12:20 PM, Matt Riddell wrote: Roger Schreiter wrote: Hi, I put a pickupgroup=0 line for each user in sip.conf. As far as I'm aware, pickup groups are only for zap interfaces... -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Pickup
On 24 Jan 2005, at 19:20, Matt Riddell wrote: Roger Schreiter wrote: Hi, I put a pickupgroup=0 line for each user in sip.conf. As far as I'm aware, pickup groups are only for zap interfaces... No, I have pickup groups working for SIP devices. As a simple thing, shouldn't the numbering for the groups start from 1? Try changing it to pickupgroup=1, thats how I have it defined for my SIP phones (Sipuras / Xlites) Phil. -- Phil Quinney IT Consultant - Any-Ideas ___ 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 Pickup
We're using them on Cisco 79XX phones without any problems, although we are using CVS-HEAD. The wiki for features.conf does mention SIP call pickup. Julian. Matt Riddell wrote: Roger Schreiter wrote: Hi, I put a pickupgroup=0 line for each user in sip.conf. As far as I'm aware, pickup groups are only for zap interfaces... ___ 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 Pickup
On 24 Jan 2005, at 19:20, Matt Riddell wrote: Roger Schreiter wrote: Hi, I put a pickupgroup=0 line for each user in sip.conf. As far as I'm aware, pickup groups are only for zap interfaces... No, I have pickup groups working for SIP devices. As a simple thing, shouldn't the numbering for the groups start from 1? Try changing it to pickupgroup=1, thats how I have it defined for my SIP phones (Sipuras / Xlites) Phil. (Apologies if this turns out as a double post...) ___ 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 Pickup
I have a rather long dial plan, but it includes support for call waiting.
However, the setgroup checkgroup commands don't seem to be working. Can
anyone help on this one?
Excerpts are below. First exten-vm is dialed and then dial-new.
As I understand, priority 1 increments the active channels for the caller
and then in dial-new priority 8 increments for Arg3, or the Callee
extension. Problem is, that priority 9 always goes on to 10 (i.e. group
never is on-the-phone.
Am I missing something?
When ext201 dials 202, CLI shows:
-- Executing Macro(SIP/201-8571, exten-vm|202|202) in new stack
-- Executing SetGroup(SIP/201-8571, 201) in new stack
-- Executing SetMusicOnHold(SIP/201-8571, default) in new stack
-- Executing SetVar(SIP/201-8571, FROMCONTEXT=exten-vm) in new stack
-- Executing GotoIf(SIP/201-8571, 0?9:5) in new stack
-- Goto (macro-exten-vm,s,5)
-- Executing Macro(SIP/201-8571, dial-new|15|tr|202|202) in new
stack
-- Executing DBget(SIP/201-8571, CallForwardIm=CF/202) in new stack
-- DBget: varname=CallForwardIm, family=CF, key=202
-- DBget: Value not found in database.
-- Executing Goto(SIP/201-8571, s|4) in new stack
-- Goto (macro-dial-new,s,4)
-- Executing DBget(SIP/201-8571, DNDStatus=DND/202) in new stack
-- DBget: varname=DNDStatus, family=DND, key=202
-- DBget: Value not found in database.
-- Executing Goto(SIP/201-8571, s|8) in new stack
-- Goto (macro-dial-new,s,8)
-- Executing SetGroup(SIP/201-8571, 202) in new stack
I'll be most grateful for any assistance.
Thanks
Mike
[macro-exten-vm]
exten = s,1,SetGroup(${CALLERIDNUM})
exten = s,2,SetMusicOnHold(default)
exten = s,3,Setvar(FROMCONTEXT=exten-vm) exten =
s,4,GotoIf($[${CALLERIDNUM} = ${ARG2}]?9:5) ;check self-voicemail exten =
s,5,Macro(dial-new,${RINGTIMER},${DIAL_OPTIONS},${ARG2},${ARG1})
[macro-dial-new]
;now check if destination is on a call
exten = s,8,SetGroup(${ARG3})
exten = s,9,CheckGroup(1)
;go to 110 then 25 if on the phone (CW handler), go to 10 if not on the
phone
exten = s,110,Goto(s,25)
;line is clear, begin dial sequence
exten = s,10,Setvar(ChanType=${E${ARG3}}) ;Get the channel type
exten = s,11,Dial(${ChanType}/${ARG3},${ARG1},${ARG2})
Mike Sander
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Re: [Asterisk-Users] Call pickup
- Original Message -
From:
Walt
Reed
To: Leandro
Cc: Walt Reed ; Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Tuesday, November 16, 2004 2:11
PM
Subject: Re: [Asterisk-Users] Call
pickup
On Tue, Nov 16, 2004 at 01:26:22PM +0100, Leandro said: From:
"Walt Reed" [EMAIL PROTECTED]
On Tue, Nov 16, 2004 at 11:04:59AM +0100, Leandro said: I
don't understand how to get call pickup to work with asterisk.
Have I to define *8 extension in the dialplan? to what?
Have I to include something, like for parked call? Has the
stable 1.0.2 version the pickup group feature? or I need to
patch it with bristuff? Search the wiki for call
pickup. It's all there. Unfortunately I have already read all
the readable on wiki without understanding the needed steps to get
call pickup to work. Can you please answer my questions?What
particular part do you not understand?The first search result hit
describes call pickup in general.The second describes how to create
pickup groups. You need to do this.The third shows where *8 is defined
and that you can change it tosomething else. *8 has been built-into
asterisk for a very long time. In1.0.2 you can change it to some other
code.That's it. Once you have defined your groups for all the
differentchannels you have (SIP, Zap, IAX, etc.), it just works. If you
haveproblems, you will need to give detailed information on how you
haveyour groups set in all the various channels involved, log examples,
etc.Make sure you look at the example configuration files that come
withasterisk.
I really hate to ask silly questions and thank you
for your time, but pickup group doesn't work yet. Maybe the pickup doesn't work
across Zap channels.
This is what I get when Zap/25 is ringing Zap/14
and Zap/7 try to pickup. I get "invalid extension" when I press *8#
- Starting simple switch on
'Zap/25-1' -- Executing Answer("Zap/25-1", "") in new
stack -- Executing Dial("Zap/25-1", "Zap/14") in new
stack -- Called 14 -- Zap/14-1 is
ringing -- Executing DigitTimeout("Zap/7-1", "3") in new
stack -- Set Digit Timeout to 3 --
Executing ResponseTimeout("Zap/7-1", "10") in new stack --
Set Response Timeout to 10 -- Zap/14-1 is
ringing -- Invalid extension '*' in context 'interno' on
Zap/7-1 == CDR updated on Zap/7-1 -- Executing
Playtones("Zap/7-1", "Congestion") in new stack -- Invalid
extension '8' in context 'interno' on Zap/7-1 == CDR updated on
Zap/7-1 -- Executing Playtones("Zap/7-1", "Congestion") in
new stack -- Invalid extension '#' in context 'interno' on
Zap/7-1 == CDR updated on Zap/7-1 -- Executing
Playtones("Zap/7-1", "Congestion") in new stack --
Zap/14-1 is ringing -- Hungup 'Zap/7-1'
This is my /etc/asterisk/zapata.conf
context=internosignalling=fxo_lsflash=100group=1callgroup=5pickupgroup=5channel
= 1-24
context=pstnsignalling=fxs_kscallgroup=5pickupgroup=5group=2channel
= 25
context=voipsignalling=fxs_kscallgroup=5pickupgroup=5group=3channel
= 26
This is the dialplan
[interno]include = parkedcalls
exten = t,1,Hangupexten =
i,1,Playtones(Congestion)
exten = s,1,DigitTimeout,3
exten = s,2,ResponseTimeout,10
exten =
4,1,Goto(componiinternoserie4,s,1)exten =
5,1,Goto(componiinternoserie5,s,1)exten =
6,1,Goto(componiinternoserie6,s,1)
exten = 0,1,Goto(impegnolinea,s,1)
exten =
3001,1,MusicOnHold()
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RE: [Asterisk-Users] Call pickup
Hi,
Have you configured features.conf file? the line which
enabled call pickup is commented and you have to un comment the line for call
pickup to work. Also you can define the numbering for call pickup
there
Thanks.
Yusuf
Alakavuk
Teknik Danman - Technical
Consultant
Grid Biliim
Teknolojileri A..
Kutepe Mahallesi Leylak
Sokak
Murat Merkezi A Blok Kat:2
Daire:9
34387 ili stanbul
Türkiye
Tel :
+90 (212) 336 92 55
Fax : +90
(212) 266 25 50
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
LeandroSent: 19 Kasm 2004 Cuma 17:52To: Walt Reed;
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users] Call pickup
- Original Message -
From:
Walt
Reed
To: Leandro
Cc: Walt Reed ; Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Tuesday, November 16, 2004 2:11
PM
Subject: Re: [Asterisk-Users] Call
pickup
On Tue, Nov 16, 2004 at 01:26:22PM +0100, Leandro said: From:
"Walt Reed" [EMAIL PROTECTED]
On Tue, Nov 16, 2004 at 11:04:59AM +0100, Leandro said: I
don't understand how to get call pickup to work with asterisk.
Have I to define *8 extension in the dialplan? to what?
Have I to include something, like for parked call? Has the
stable 1.0.2 version the pickup group feature? or I need to
patch it with bristuff? Search the wiki for call
pickup. It's all there. Unfortunately I have already read all
the readable on wiki without understanding the needed steps to get
call pickup to work. Can you please answer my questions?What
particular part do you not understand?The first search result hit
describes call pickup in general.The second describes how to create
pickup groups. You need to do this.The third shows where *8 is defined
and that you can change it tosomething else. *8 has been built-into
asterisk for a very long time. In1.0.2 you can change it to some other
code.That's it. Once you have defined your groups for all the
differentchannels you have (SIP, Zap, IAX, etc.), it just works. If you
haveproblems, you will need to give detailed information on how you
haveyour groups set in all the various channels involved, log examples,
etc.Make sure you look at the example configuration files that come
withasterisk.
I really hate to ask silly questions and thank you
for your time, but pickup group doesn't work yet. Maybe the pickup doesn't work
across Zap channels.
This is what I get when Zap/25 is ringing Zap/14
and Zap/7 try to pickup. I get "invalid extension" when I press *8#
- Starting simple switch on
'Zap/25-1' -- Executing Answer("Zap/25-1", "") in new
stack -- Executing Dial("Zap/25-1", "Zap/14") in new
stack -- Called 14 -- Zap/14-1 is
ringing -- Executing DigitTimeout("Zap/7-1", "3") in new
stack -- Set Digit Timeout to 3 --
Executing ResponseTimeout("Zap/7-1", "10") in new stack --
Set Response Timeout to 10 -- Zap/14-1 is
ringing -- Invalid extension '*' in context 'interno' on
Zap/7-1 == CDR updated on Zap/7-1 -- Executing
Playtones("Zap/7-1", "Congestion") in new stack -- Invalid
extension '8' in context 'interno' on Zap/7-1 == CDR updated on
Zap/7-1 -- Executing Playtones("Zap/7-1", "Congestion") in
new stack -- Invalid extension '#' in context 'interno' on
Zap/7-1 == CDR updated on Zap/7-1 -- Executing
Playtones("Zap/7-1", "Congestion") in new stack --
Zap/14-1 is ringing -- Hungup 'Zap/7-1'
This is my /etc/asterisk/zapata.conf
context=internosignalling=fxo_lsflash=100group=1callgroup=5pickupgroup=5channel
= 1-24
context=pstnsignalling=fxs_kscallgroup=5pickupgroup=5group=2channel
= 25
context=voipsignalling=fxs_kscallgroup=5pickupgroup=5group=3channel
= 26
This is the dialplan
[interno]include = parkedcalls
exten = t,1,Hangupexten =
i,1,Playtones(Congestion)
exten = s,1,DigitTimeout,3
exten = s,2,ResponseTimeout,10
exten =
4,1,Goto(componiinternoserie4,s,1)exten =
5,1,Goto(componiinternoserie5,s,1)exten =
6,1,Goto(componiinternoserie6,s,1)
exten = 0,1,Goto(impegnolinea,s,1)
exten =
3001,1,MusicOnHold()
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Re: [Asterisk-Users] Call pickup
- Original Message - From: Yusuf Alakavuk To: 'Asterisk Users Mailing List - Non-Commercial Discussion' ; 'Walt Reed' Sent: Friday, November 19, 2004 5:02 PM Subject: RE: [Asterisk-Users] Call pickup Hi, Have you configured features.conf file? the line which enabled call pickup is commented and you have to un comment the line for call pickup to work. Also you can define the numbering for call pickup there Are you referring to pickupexten=*8? Thank you for your try, but unfortunately, I have already uncommented it in features.conf:-( ;; Sample Parking configuration; [general]parkext = 700 ; What ext. to dial to parkparkpos = 701-720 ; What extensions to park calls oncontext = 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;adsipark = yes ; if you want ADSI parking announcements pickupexten = *8 ___ 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 pickup - Adding more info on my pickup weirdness.
Adding more info on my pickup weirdness, I try other "embedded extensions", like *70 or *69. No embedded extensions are working. Asterisk version is stable 1.0.2. Channels are Zap via channel bank and a T100P. Leandro ___ 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 pickup
On Tue, Nov 16, 2004 at 11:04:59AM +0100, Leandro said: I don't understand how to get call pickup to work with asterisk. Have I to define *8 extension in the dialplan? to what? Have I to include something, like for parked call? Has the stable 1.0.2 version the pickup group feature? or I need to patch it with bristuff? Search the wiki for call pickup. It's all there. ___ 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 pickup
- Original Message - From: Walt Reed [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Tuesday, November 16, 2004 1:04 PM Subject: Re: [Asterisk-Users] Call pickup On Tue, Nov 16, 2004 at 11:04:59AM +0100, Leandro said: I don't understand how to get call pickup to work with asterisk. Have I to define *8 extension in the dialplan? to what? Have I to include something, like for parked call? Has the stable 1.0.2 version the pickup group feature? or I need to patch it with bristuff? Search the wiki for call pickup. It's all there. Unfortunately I have already read all the readable on wiki without understanding the needed steps to get call pickup to work. Can you please answer my questions? Thank you Leandro ___ 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 pickup
On Tue, Nov 16, 2004 at 01:26:22PM +0100, Leandro said: From: Walt Reed [EMAIL PROTECTED] On Tue, Nov 16, 2004 at 11:04:59AM +0100, Leandro said: I don't understand how to get call pickup to work with asterisk. Have I to define *8 extension in the dialplan? to what? Have I to include something, like for parked call? Has the stable 1.0.2 version the pickup group feature? or I need to patch it with bristuff? Search the wiki for call pickup. It's all there. Unfortunately I have already read all the readable on wiki without understanding the needed steps to get call pickup to work. Can you please answer my questions? What particular part do you not understand? The first search result hit describes call pickup in general. The second describes how to create pickup groups. You need to do this. The third shows where *8 is defined and that you can change it to something else. *8 has been built-into asterisk for a very long time. In 1.0.2 you can change it to some other code. That's it. Once you have defined your groups for all the different channels you have (SIP, Zap, IAX, etc.), it just works. If you have problems, you will need to give detailed information on how you have your groups set in all the various channels involved, log examples, etc. Make sure you look at the example configuration files that come with asterisk. ___ 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 pickup
On Tue, Nov 16, 2004 at 11:04:59AM +0100, Leandro said: I don't understand how to get call pickup to work with asterisk. Have I to define *8 extension in the dialplan? to what? Have I to include something, like for parked call? Has the stable 1.0.2 version the pickup group feature? or I need to patch it with bristuff? Search the wiki for call pickup. It's all there. Unfortunately I have already read all the readable on wiki without understanding the needed steps to get call pickup to work. Can you please answer my questions? It really isn't that hard. Here's an example. In zapata.conf, an entry might look like: context-inbound-bus signalling=fxs_ks snip other detail callgroup=2 channel = 1 In sip.conf, an phone entry might look like: [3002] type= username=3002 secret= snip other detail pickupgroup=2 Since the above reflects a zap interface was assigned to callgroup=2, the sip phone with pickupgroup=2 can pick that ringing call up by pressing *8 (or *8#). If a different sip phone is defined with pickupgroup=17, it would not be able to get callgroup=2 assignments. To take that a step further, you could also have a sip.conf entry like: [3004] type= username=3004 secret= pickupgroup=2 callgroup=2 and whenever x3004 is ringing, the sip phone at 3002 can pick that ringing call up as well as the zap interface noted above. If both are ringing at exactly the same time, I'm not sure which will be picked up, but one of them will be. On my sip phone (Cisco 7960) I have to use *8# to pickup calls. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Pickup
On Thu, Nov 11, 2004 at 07:57:11PM -0500, Jerry Geis said: On my present phone system I can pickup a call that is ringing on another phone. How do I do this with asterisk? I searched on the wiki for pickup and did not find anything. Hmm. I just did a search on call pickup on the wiki and it had 547 results. The first hit mentioned *8 in Asterisk. The second hit showed how to configure groups. The third was features.conf which shows that you can change *8 to some other code. Are you looking at the right wiki http://www.voip-info.org/ ___ 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 Pickup
On Thu, 2004-11-11 at 19:57 -0500, Jerry Geis wrote: On my present phone system I can pickup a call that is ringing on another phone. How do I do this with asterisk? I searched on the wiki for pickup and did not find anything. pickupgroups/callgroups -- Steven Critchfield [EMAIL PROTECTED] ___ 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
[Fwd: Re: [Asterisk-Users] call pickup fails.]
More than one hundred messages related to *8 or call pickup problem in last 6 months!! Please someone in the development team could clarify this and make himself responsible for the response. By now It seems a bad joke. We have spent thousand dollars with hardware, sip phones, working men hours, and with digium stuff (E1, fxo, fxs cards etc) and we have had the *8 problem (sip callee ringing forever) al least for 6 months. This made us to lose at least a couple of clients (a IP PBX where you are not able to pickup correctly other SIP extensions, are you fooling, come back next year ) an we keep reading again and again people saying it is not working, and a couple of enlighted people saying their have the luck to have it working!! Please this is not serious! This should be fixed for every-one-of-us (if you are one of the lucky boys send a sip.conf to THIS LIST or post it in wiki-asterisk with a couple of client definitions where people from the earth will be able to pick up it) or be recogniced as not working (most of the time if you prefer) and ask for someone to solve it (as an open bug report for example). Is not so complicated stuff to put a callgroup=1 an a pickupgroup=1 in a file to suspect we are all fools not getting it to work because of some sort of mental illness, or I'm wrong. If someone feels himself intelligent by this, he have a problem!! The money we have invested in Digium and Asterisk stuff in the last six months is the same money half of the people in my country has to live eighteen years!! More or less 450 times our basic salary here, so: Please, there is people betting on open source software and loosing money out there because of these funny details, and that's the same people making Digium earn their bucks. Sorry for my bad (o I should say mad?) english :( Thanks for your attention guys Luis Pd: despite *8 pickup, asterisk is great (most of the time) :) Original Message Subject:Re: [Asterisk-Users] call pickup fails. Date: Thu, 27 May 2004 07:38:44 -0600 From: Rich Adamson [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] References: [EMAIL PROTECTED] I saw a few weeks ago a discussion about cal pickup, *8, not working but did not find a message about it being resolved, I look for a bug on the bug list but did not find anything about it not working, nor a bug open. I installed asterisk 0.9.0, have one sip fxo gateway and only sip phones, all of them have callgroup=1 and pickupgroup=1 but I can not get a call that is ringing in another phone, there is a message on the * console that says something like Nothing to pickup every time I try it. Any hints ? It's been working fine for me on cvs Head for months. We have to use *8# from a sip phone however. ___ 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: [Fwd: Re: [Asterisk-Users] call pickup fails.]
More than one hundred messages related to *8 or call pickup problem in last 6 months!! Please someone in the development team could clarify this and make himself responsible for the response. I'm not sure what you're asking for, but *8# has been working just fine here since about October last year and still working fine on current Head cvs. If you're asking for something else, then how about rewording it. If you really are talking about plain old call pickup, our cisco 7960's work just fine with a sip.conf entry like: [3001] type=friend username=3001 secret=mysecret host=dynamic context=sip-in callgroup=2 pickupgroup=2 mailbox=3001 with extensions.conf entries like: exten = 3002,1,Dial(SIP/3002,15) exten = 3002,2,Voicemail2(u3002) exten = 3002,102,Voicemail2(b3002) exten = 3002,103,Hangup and incoming fxo lines in zapata.conf like: context=inbound-bus snip callgroup=2 channel = 4 If that's what you want and it isn't working, then I'd suggest reviewing your dialplan. ___ 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 pickup fails.
I saw a few weeks ago a discussion about cal pickup, *8, not working but did not find a message about it being resolved, I look for a bug on the bug list but did not find anything about it not working, nor a bug open. I installed asterisk 0.9.0, have one sip fxo gateway and only sip phones, all of them have callgroup=1 and pickupgroup=1 but I can not get a call that is ringing in another phone, there is a message on the * console that says something like Nothing to pickup every time I try it. Any hints ? It's been working fine for me on cvs Head for months. We have to use *8# from a sip phone however. ___ 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 pickup - phone continues to ring - still a problem?
I am still experiencing the problem where you pick up an incoming analog call ringing on SIP Phone A with SIP Phone B using *8 but Phone A continues to ring. This happens with Grandstreams and Snoms on the 0.7.1 code base. My theory is that Asterisk is not telling Phone A to stop ringing when the pickup occurs but I don't really know. The problem does not occur when it is purely a SIP-to-SIP phone call. Does anyone have a solution? ___ 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 pickup - phone continues to ring - still a problem?
Il 01:02, venerdì 02 aprile 2004, John Vogel ha scritto: I am still experiencing the problem where you pick up an incoming analog call ringing on SIP Phone A with SIP Phone B using *8 but Phone A continues to ring. This happens with Grandstreams and Snoms on the 0.7.1 code base. My theory is that Asterisk is not telling Phone A to stop ringing when the pickup occurs but I don't really know. The problem does not occur when it is purely a SIP-to-SIP phone call. Does anyone have a solution? How about callgroups and pickupgroup in sip.conf? ___ 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 pickup via *8 from ata186 (SIP)
It works for me with sip 2.15, 2.16.x and 3 versions. - Original Message - From: Anton Yurchenko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, December 29, 2003 6:42 AM Subject: [Asterisk-Users] call pickup via *8 from ata186 (SIP) Hello, Does call pickup works with ATA-186 SIP? at the same pbx it works with MGCP but bit ata-186 with SIP it doesnt work, just nothing happens. Anyone have it working? Also it seems that when typing reload on the console, the asterisk doesnt reread the mgcp.conf. please answer Thanks -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call pickup via *8 from ata186 (SIP)
CW_ASN wrote: It works for me with sip 2.15, 2.16.x and 3 versions. I have FW version : *ata0009e88e33cd* Version: v2.15 ata18x (Build 020927a) MAC: 0.9.232.142.51.205 what other parameters should I use? What AudioMode? thanks - Original Message - From: Anton Yurchenko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, December 29, 2003 6:42 AM Subject: [Asterisk-Users] call pickup via *8 from ata186 (SIP) Hello, Does call pickup works with ATA-186 SIP? at the same pbx it works with MGCP but bit ata-186 with SIP it doesnt work, just nothing happens. Anyone have it working? Also it seems that when typing reload on the console, the asterisk doesnt reread the mgcp.conf. please answer Thanks -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call pickup via *8 from ata186 (SIP)
CW_ASN wrote: Actually, I'm using: Thanks a lot , I`ll try this Version: v3.0.0 atasip (Build 031210A) AudioMode: 0x00140014 CallFeatures: 0x PadiFeatures: 0x CallerIdMethod: 0x00019e60 FeatureTimer: 0x FeatureTimer2: 0x001e Polarity: 0x ConnectMode: 0x00460400 TimeZone: 17 TOS: 0x00a0 SigTimer: 0x01418564 OpFlags: 0x0002 VLANSetting: 0x002b Please re-check your sip.conf configuration, make sure that pickupgroup and callgroup are present and correctly referenced with another device. Hope this helps Regards, Gustavo - Original Message - From: Anton Yurchenko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, December 29, 2003 8:30 AM Subject: Re: [Asterisk-Users] call pickup via *8 from ata186 (SIP) CW_ASN wrote: It works for me with sip 2.15, 2.16.x and 3 versions. I have FW version : *ata0009e88e33cd* Version: v2.15 ata18x (Build 020927a) MAC: 0.9.232.142.51.205 what other parameters should I use? What AudioMode? thanks - Original Message - From: Anton Yurchenko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, December 29, 2003 6:42 AM Subject: [Asterisk-Users] call pickup via *8 from ata186 (SIP) Hello, Does call pickup works with ATA-186 SIP? at the same pbx it works with MGCP but bit ata-186 with SIP it doesnt work, just nothing happens. Anyone have it working? Also it seems that when typing reload on the console, the asterisk doesnt reread the mgcp.conf. please answer Thanks -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call pickup via *8 from ata186 (SIP)
Hi! Also it seems that when typing reload on the console, the asterisk doesnt reread the mgcp.conf. That's correct, unfortunately - see the MGCP section at bugs.digium.com or do a help reload on the CLI to learn more. Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call pickup (*8) on SIP devices.
Just submitted a patch for this on asterisk-dev. Quick fix add the following line above line 5022 in chan_sip.c ast_setstate(c,AST_STATE_DOWN); Just updated to current cvs a few minutes ago primarily to get the call pickup to function properly. Using C7960's and Snom 200 on RH9. All compiled and installed cleanly. Maybe I'm misunderstanding the call pickup functions; here's a couple of samples from my sip.conf: [3000] type=friend username=3000 secret=mypassword host=dynamic context=from-sip callgroup=2 pickupgroup=2 mailbox=3000 [3001] type=friend username=3001 secret=mypassword2 host=dynamic context=from-sip callgroup=2 pickupgroup=2 callgroup=2 mailbox=3001 [3002] type=friend username=3002 secret=mypassword3 host=dynamic context=from-sip callgroup=2 pickupgroup=2 mailbox=3002 If station 3002 calls 3001, I'm expecting the user at 3000 to hear the rining at 3001, and dial *8# to pick it up. When I try that, *8# does not pick up the call and only receives a busy. Are my expectations incorrect, my definitions, or what? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call pickup (*8) on SIP devices.
Did you try to use *8 only instead of *8# ? Last time when I tried *8 picked the call with known results but I haven't tested any patches yet. I really hope call pickup now works. -- Pertti Rich Adamson wrote: Just submitted a patch for this on asterisk-dev. Quick fix add the following line above line 5022 in chan_sip.c ast_setstate(c,AST_STATE_DOWN); Just updated to current cvs a few minutes ago primarily to get the call pickup to function properly. Using C7960's and Snom 200 on RH9. All compiled and installed cleanly. Maybe I'm misunderstanding the call pickup functions; here's a couple of samples from my sip.conf: [3000] type=friend username=3000 secret=mypassword host=dynamic context=from-sip callgroup=2 pickupgroup=2 mailbox=3000 [3001] type=friend username=3001 secret=mypassword2 host=dynamic context=from-sip callgroup=2 pickupgroup=2 callgroup=2 mailbox=3001 [3002] type=friend username=3002 secret=mypassword3 host=dynamic context=from-sip callgroup=2 pickupgroup=2 mailbox=3002 If station 3002 calls 3001, I'm expecting the user at 3000 to hear the rining at 3001, and dial *8# to pick it up. When I try that, *8# does not pick up the call and only receives a busy. Are my expectations incorrect, my definitions, or what? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users - ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call pickup (*8) on SIP devices.
Just submitted a patch for this on asterisk-dev.
Quick fix add the following line above line 5022 in chan_sip.c
ast_setstate(c,AST_STATE_DOWN);
Should look like this when you are done.
} else {
5021ast_mutex_unlock(p-lock);
5022ast_setstate(c,
AST_STATE_DOWN);
5023ast_hangup(c);
5024ast_mutex_lock(p-lock);
c = NULL;
-Scott
-Original Message-
From: James Sizemore [mailto:[EMAIL PROTECTED]
Sent: Thursday, October 23, 2003 2:04 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Call pickup (*8) on SIP devices.
Yes
Ing. Angel Gomez Garcia wrote:
Hello.
I have this issue, when I pickup a call that is ringing in a SIP
Phone, it keeps ringing.
There is bug #116 that mention something about these, but it does
not seem to be resolved , at least, not yet.
Anybody else has seen it behavior ?
Thank's.
___
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Re: [Asterisk-Users] Call pickup (*8) on SIP devices.
Bisker, Scott (7805) wrote: Just submitted a patch for this on asterisk-dev GGrreeaatt!! Will test ASAP. Thank's. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call pickup (*8) on SIP devices.
Ing. Angel Gomez Garcia wrote: Hello. I have this issue, when I pickup a call that is ringing in a SIP Phone, it keeps ringing. There is bug #116 that mention something about these, but it does not seem to be resolved , at least, not yet. Anybody else has seen it behavior ? Thank's. Everyone.. :) Its a known issue.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call pickup (*8) on SIP devices.
WipeOut wrote: Ing. Angel Gomez Garcia wrote: Hello. I have this issue, when I pickup a call that is ringing in a SIP Phone, it keeps ringing. There is bug #116 that mention something about these, but it does not seem to be resolved , at least, not yet. Anybody else has seen it behavior ? Thank's. Everyone.. :) Its a known issue.. Later.. OOhh :( Any known workaround ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call pickup (*8) on SIP devices.
Ing. Angel Gomez Garcia wrote: WipeOut wrote: Ing. Angel Gomez Garcia wrote: Hello. I have this issue, when I pickup a call that is ringing in a SIP Phone, it keeps ringing. There is bug #116 that mention something about these, but it does not seem to be resolved , at least, not yet. Anybody else has seen it behavior ? Thank's. Everyone.. :) Its a known issue.. Later.. OOhh :( Any known workaround ? Not that I know of.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call pickup (*8) on SIP devices.
WipeOut wrote: Ing. Angel Gomez Garcia wrote: WipeOut wrote: Ing. Angel Gomez Garcia wrote: Hello. I have this issue, when I pickup a call that is ringing in a SIP Phone, it keeps ringing. There is bug #116 that mention something about these, but it does not seem to be resolved , at least, not yet. Anybody else has seen it behavior ? Thank's. Everyone.. :) Its a known issue.. Later.. OOhh :( Any known workaround ? Not that I know of.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Here with a snom200/SIP and ATA-186/MGCP everything works fine (i dial *8 to pick up a call). -Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call pickup (*8) on SIP devices.
Thomas Dingermann wrote: WipeOut wrote: Ing. Angel Gomez Garcia wrote: WipeOut wrote: Ing. Angel Gomez Garcia wrote: Hello. I have this issue, when I pickup a call that is ringing in a SIP Phone, it keeps ringing. There is bug #116 that mention something about these, but it does not seem to be resolved , at least, not yet. Anybody else has seen it behavior ? Thank's. Everyone.. :) Its a known issue.. Later.. OOhh :( Any known workaround ? Not that I know of.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Here with a snom200/SIP and ATA-186/MGCP everything works fine (i dial *8 to pick up a call). -Thomas May be becasue you are using SIP and MGCP.. But when using 2 SIP UA's the phone definately keeps on ringing.. :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call pickup (*8) on SIP devices.
Here are some ideas for anyone with some extra time on there hands. SIP phones on call pickup either use a special REGISTER or you can place a call with the magic extension and have the switch hang up on you and immediately call you back. With the second option, you could dial *8, Asterisk could check to see if this was a SIP channel and if so, hangup on the call and somehow add that channel to the current list of ringing phones. From the knowledge I have of Asterisk which basically is configuration info only I'm not sure if this is a clean approach with the current architecture. In the commercial SIP world, a forking proxy would be used and the pickup phone's contact would be added to the ringing phones contact list for that call only. The proxy would fork an additional INVITE to the pickup phone and both phones would ring. That way the pickup phone or the original destination phone could answer the call. Maybe these ideas will spark some coding. :) WipeOut wrote: Ing. Angel Gomez Garcia wrote: WipeOut wrote: Ing. Angel Gomez Garcia wrote: Hello. I have this issue, when I pickup a call that is ringing in a SIP Phone, it keeps ringing. There is bug #116 that mention something about these, but it does not seem to be resolved , at least, not yet. Anybody else has seen it behavior ? Thank's. Everyone.. Its a known issue.. Later.. OOhh Any known workaround ? Not that I know of.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call pickup (*8) on SIP devices.
okay someone find me on IRC where I can ssh in and i'll really try to fix this. Mark On Thu, 23 Oct 2003, WipeOut wrote: Ing. Angel Gomez Garcia wrote: Hello. I have this issue, when I pickup a call that is ringing in a SIP Phone, it keeps ringing. There is bug #116 that mention something about these, but it does not seem to be resolved , at least, not yet. Anybody else has seen it behavior ? Thank's. Everyone.. :) Its a known issue.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call pickup (*8) on SIP devices.
Yes Ing. Angel Gomez Garcia wrote: Hello. I have this issue, when I pickup a call that is ringing in a SIP Phone, it keeps ringing. There is bug #116 that mention something about these, but it does not seem to be resolved , at least, not yet. Anybody else has seen it behavior ? Thank's. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Pickup problem with cisco 7960 (SIP)
On Mon, 2003-09-22 at 15:42, Manuel Marn Garca wrote: Please help! When I try to place a call pickup from a cisco phone 7960 using *8 the call is picked up but the other phone continues ringing. Is there any problem with call pickup in SIP. It's a known problem... I wish someone would hurry up and fix it. Jared Smith ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Pickup problem with cisco 7960 (SIP)
Here's one thats way out in left field... don't use call pickup! :P Problem solved sorta! bkw On Mon, 22 Sep 2003, Jared Smith wrote: On Mon, 2003-09-22 at 15:42, Manuel Marn Garca wrote: Please help! When I try to place a call pickup from a cisco phone 7960 using *8 the call is picked up but the other phone continues ringing. Is there any problem with call pickup in SIP. It's a known problem... I wish someone would hurry up and fix it. Jared Smith ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Pickup
You need to have a pending call in the system (some extensions that is ringing to test that). If you have 3 FXS ports try to place a call from the first one to the 2nd and then instead of taking the 2nd off hook dial *8 on the 3rd phone Martin On Thu, 17 Jul 2003, Jay Tyndall wrote: Hi, I have been trying to workout how to use the call pickup. So Far, I have the following in zapata.conf [channels] signalling = fxo_ks context = local pickupgroup=1 callgroup=1 channel = 1-3 When I dial *8# all I hear is busy tone. What have I missed? thanks Jay. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
