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: 3c953745720b
[asterisk-users] Call pickup on channel sip with SNOM phones issue
Hi, while trying to get my new Asterisk 15.5.0 PBX replacing a 11 years old Asterisk 1.2.31 ISDN BPX, I'm stuck to get call pickup going as usual. The old one uses specific patches, IIRC... If I interpret various sources of related information correctly, current Asterisk versions should support this feature out of the box. According to http://wiki.snom.com/Category:HowTo:Call_Pickup, there are several ways to get this feature going. I'm enjoying method (1) since ages, but I couldn't get asterisk to send the full NOTIFY xml dialog-info including call-id, remote and local values, although setting context = phones allowsubscribe = yes subscribecontext = phones notifyringing = yes notifycid = ignore-context as well as callgroup = 1 pickupgroup = 1 for every local phone (all snom, mostly 360 phones) in sip.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) but the notify looks like this: --- == Extension Changed 62[phones] new state Ringing for Notify User 62 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=z9hG4bK7d7c5ac4;rport Max-Forwards: 70 From: ;tag=as72f9c98b To: ;tag=ufln5vo7x5 Contact: Call-ID: 3c94f02212d4-8we7ggt625fi CSeq: 106 NOTIFY User-Agent: Asterisk PBX 15.5.0 Subscription-State: active Event: dialog Content-Type: application/dialog-info+xml Content-Length: 217 early <-> --- (9 headers 0 lines) --- <--- 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=z9hG4bK7d7c5ac4;rport=5060 From: ;tag=as72f9c98b To: ;tag=ufln5vo7x5 Call-ID: 3c94f02212d4-8we7ggt625fi CSeq: 106 NOTIFY Content-Length: 0 SIP/60 is notified correctly, but misses the notifycid information. I've tried both, notifycid = yes and notifycid = ignore-context of course. *CLI> core show hints 62@phones : SIP/62State:Idle Presence:not_set Watchers 3 61@phones : SIP/61State:Idle Presence:not_set Watchers 2 60@phones : SIP/60State:Idle Presence:not_set Watchers 3 *CLI> sip show subscriptions Peer User Call ID ExtensionLast state TypeMailboxExpiry 172.16.23.60 60 3c94f0220769-0p 61@phonesIdle dialog-info+xml 003600 172.16.23.60 60 3c94f0220182-27 60@phonesIdle dialog-info+xml 003600 172.16.23.62 62 313533353338323 60@phonesIdle dialog-info+xml 003600 172.16.23.62 62 313533353338323 62@phonesIdle dialog-info+xml 003600 172.16.23.60 60 3c94f0220d01-p2 62@phonesIdle dialog-info+xml 003600 172.16.23.60 60 3c94f023462b-j2 60@phonesIdle dialog-info+xml 003600 172.16.23.60 60 3c94f02212d4-8w 62@phonesIdle dialog-info+xml 003600 172.16.23.62 62 313533353338323 61@phonesIdle dialog-info+xml 003600 What am I missing here, any suggestions? Cheers, Pete -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started 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
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 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
Check out connectedline() -Original Message- From: Rusty Newton Sender: asterisk-users-boun...@lists.digium.com Date: Tue, 19 Feb 2013 09:58:30 To: Asterisk Users Mailing List - Non-Commercial Discussion Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Pickup how to display CND of incoming number - Original Message - > From: "David C Klaverstyn" > 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=1&t=71351&p=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 how to display CND of incoming number
- Original Message - > From: "David C Klaverstyn" > 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=1&t=71351&p=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
[asterisk-users] Call Pickup how to display CND of incoming number
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-- _ -- 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 mailto: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?
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
[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
Re: [asterisk-users] call pickup
Search for dialog-info pickup -Original Message- From: Marek Cervenka Sender: asterisk-users-boun...@lists.digium.com Date: Fri, 07 Oct 2011 09:47:45 To: Asterisk Users Mailing List - Non-Commercial Discussion Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion 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
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
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. 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 :-) Regards Martin -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] call pickup
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? thanks -- --- Marek Cervenka === -- _ -- 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: sip:35...@10.40.23.179 sip:35...@10.40.23.179 early 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" ;tag=o28fq65rfu To: "Lab 1" Call-ID: 3c2672b3f35a-dpd0zv11yegl CSeq: 1 INVITE Max-Forwards: 70 Contact: ;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: Totag: [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: Totag: [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 -- 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 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 wrote:
From: Loris Santamaria
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:
sip:35...@10.40.23.179
sip:35...@10.40.23.179
early
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" ;tag=o28fq65rfu
To: "Lab 1"
Call-ID: 3c2672b3f35a-dpd0zv11yegl
CSeq: 1 INVITE
Max-Forwards: 70
Contact: ;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: Totag:
[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: Totag:
[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
[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: sip:35...@10.40.23.179 sip:35...@10.40.23.179 early 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" ;tag=o28fq65rfu To: "Lab 1" Call-ID: 3c2672b3f35a-dpd0zv11yegl CSeq: 1 INVITE Max-Forwards: 70 Contact: ;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: Totag: [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: Totag: [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
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
[asterisk-users] Call Pickup Works w/Linksys ATA, not with Cisco 7940G
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) What could the difference be? Below you will find: (a) the "sip show peer nnn" for an ATA extension and a Cisco extension (b) the SIP debug trace for (i) a successful call pickup from the ATA and (ii) an unsuccessful call pickup from the Cisco Any light anyone can shed on the perplexing problem would be most appreciated. I have a forehead-shaped dent in the wall that is growing larger. Linksys SPA2102 ATA "sip show peer 101": * Name : 101 Secret : MD5Secret: Context : numberplan-custom-1 Subscr.Cont. : Language : AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: 1 Pickupgroup : 1 Mailbox : 101 VM Extension : asterisk LastMsgsSent : 0/3 Call limit : 0 Dynamic : Yes Callerid : "Dxx Gxxx" <604-123-> MaxCallBR: 384 kbps Expire : 1861 Insecure : no Nat : RFC3581 ACL : No T38 pt UDPTL : No CanReinvite : No PromiscRedir : No User=Phone : No Video Support: No Trust RPID : No Send RPID: No Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr->IP : 192.168.0.205 Port 5060 Defaddr->IP : 0.0.0.0 Port 5060 Def. Username: 101 SIP Options : replaces replace Codecs : 0x8000e (gsm|ulaw|alaw|h263) Codec Order : (none) Auto-Framing: No Status : OK (5 ms) Useragent: Linksys/SPA2102-5.2.3 Reg. Contact : sip:1...@192.168.0.205:5060 Cisco 7940G Phone "sip show peer 106": * Name : 106 Secret : MD5Secret: Context : numberplan-custom-1 Subscr.Cont. : Language : AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: 1 Pickupgroup : 1 Mailbox : 106 VM Extension : asterisk LastMsgsSent : 3/1 Call limit : 0 Dynamic : Yes Callerid : "Cxxx Nxxx" <6041234567> MaxCallBR: 384 kbps Expire : 247 Insecure : no Nat : RFC3581 ACL : No T38 pt UDPTL : No CanReinvite : No PromiscRedir : No User=Phone : No Video Support: No Trust RPID : No Send RPID: No Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr->IP : 192.168.0.211 Port 5060 Defaddr->IP : 0.0.0.0 Port 5060 Def. Username: 106 SIP Options : (none) Codecs : 0x8000e (gsm|ulaw|alaw|h263) Codec Order : (none) Auto-Framing: No Status : OK (195 ms) Useragent: Cisco-CP7940G/8.0 Reg. Contact : sip:1...@192.168.0.211:5060;transport=udp Successful *8 Call Pickup (SIP Trace) <--- SIP read from 192.168.0.205:5060 ---> INVITE sip:*...@192.168.0.12 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.205:5060;branch=z9hG4bK-e7b4459c From: 101 ;tag=22b459c8b65178bco0 To: Remote-Party-ID: 101 ;screen=yes;party=calling Call-ID: 94dc7b8-591d6...@192.168.0.205 CSeq: 101 INVITE Max-Forwards: 70 Contact: 101 Expires: 240 User-Agent: Linksys/SPA2102-5.2.3 Content-Length: 444 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces Content-Type: application/sdp v=0 o=- 922981 922981 IN IP4 192.168.0.205 s=- c=IN IP4 192.168.0.205 t=0 0 m=audio 16412 RTP/AVP 0 2 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-> --- (15 headers 20 lines) --- Sending to 192.168.0.205 : 5060 (no NAT) Using INVITE request as basis request - 94dc7b8-591d6...@192.168.0.205 Found peer '101' tg2*CLI> <--- Reliably Transmitting (no NAT) to 192.168.0.205:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.205:5060;branch=z9hG4bK-e7b4459c;received=192.168.0.205 From: 101 ;tag=22b459c8b65178bco0 To: ;tag=as4bf7113a Call-ID: 94dc7b8-591d6...@192.168.0.205 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="637bf838" Content-Length: 0 <> Scheduling destruction of SIP dialog '94dc7b8-591d6...@192.168.0.205' in 6400 ms (Method: INVITE) tg2*CLI> <--- SIP read from 192.168.0.205:5060 ---> ACK sip:*...@192.168.0.12 S
[asterisk-users] Call pickup with IAX
Hi all Somebody know with IAX support pickup call feature in the last 1.4 .X asterisk release ? With SIP I use features.conf and works fine, but no way to make works with IAX. Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call pickup and ring groups
--- On Fri, 3/6/09, Vieri 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.
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[asterisk-users] call pickup and ring groups
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
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[asterisk-users] Call Pickup (*8) / Attended forward and CallerID
Hi, Since we're moving from a legacy PABX that has been serving one of our customers for more than 15 years, we'd like this process to require no "human habits" change among the users. Software: Asterisk 1.4.22 Hardware: Polycom phones (mainly 430/601) Here are the "problems": We did configure call groups, pickup groups, ... - When someone picks up a call from another person, the display of his phone only shows *8 and not the original CallerID. - When doing an attended transfer, the callerid of the original caller (A calls B, then B forwards to C => We want to show C the original callerid somewhere on his phone's screen). - When using the blind transfer feature, the CallerID is fine. I know this has already been discussed in 2006 (from digium's BTS), and would like to know if this situation did change, or not. Is it still considered as features ? Is it considered as bugs ? Will it be implemented in another way in some future release ? ...? Thanks Laurent ___ -- 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 - 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
[asterisk-users] call pickup - Asterisk 1.4.19.1 -
Hello List, Does anyone here have call pickup (with *8 ) working ok on Asterisk version 1.4.19.1 ? Thanks in advice, -- Jose P. Espinal Slackware-Es.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] call pickup problem
i have TB instaled and i cant get call pickup when another phone rings i tried ** , *8 , *8# , **+ext but nothing seems to be ok.on extention menu i put call pickup=1 and call group=1 but nothing look at my features.conf; ; Sample Parking configuration ; [general] ; do not manually enter parkinglot config information, use the parkinglot module ; ; the parking_additional.inc file is auto-generated by the Parkinglot Module, do ; not hand edit that file #include parking_additional.inc #include features_general_custom.conf [applicationmap] #include features_applicationmap_additional.conf ; *** IMPORTANT NOTE *** ; The original blindxfer was '#', and has been changed to '##' to avoid ; issues with sending DTMF '#' to remote parties. [featuremap] blindxfer => ##; Blind Transfer disconnect => **; Disconnect Call automon => *1; One Touch Record ;atxfer => *2; Attended Xfer please tell the right steps for make it working thank you ___ --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(extension1&extension2&...) > > 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
[asterisk-users] Call Pickup with more than one argument
Dear all, Does Pickup application accept multiple extensions pickup syntax, like the following line? Pickup(extension1&extension2&...) 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? Thanks, Ricardo. ___ --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 from >> >> 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 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
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 from 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
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 from 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
[Asterisk-Users] Call-pickup function in Queue application
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
[Asterisk-Users] Call Pickup with CID info
I was wondering if someone could help me with this, I’ve searched high and low to find more info but with no success. When I want to transfer a call from another ringing sip phone to my sip handset I dial *8. This works but the caller ID shows up as *8 on my handset. What I want to do is be able to have the original caller id come up on my SIP phone rather than *8. There was a page on VOIP info (http://www.voip-info.org/wiki/view/Asterisk+bounty+Call+Pickup+with+CID+info.+-+SIP ) that talked about doing this but it doesn’t seem to be available anymore. Any suggestions or help would be greatly appreciated. Regards, Bevan ___ --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
Cheers Doug, Thank you all for the help. I'll upgrade to 1.2.5 soon. Much appreciated! Thanks to all who contributed! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Monday, 20 March 2006 11:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: 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 ___ --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
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
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
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 > > > > > > 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: > > > > > > > ___ --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 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: ___ --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 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: -- 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
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
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
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: -- 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
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
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
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'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
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
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
[Asterisk-Users] Call Pickup Woes
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
[Asterisk-Users] Call pickup between different protocols
Hi, I'm unable to pickup a call (*8) directed to a SIP phone from a IAX2 phone. Is it normal? I don't see ant pickupgroup/callgroup setting in iax.conf... -- Domenico Viggiani ___ --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
[Asterisk-Users] Call Pickup in [EMAIL PROTECTED]
Anyone out there knows how the call-pickup works on [EMAIL PROTECTED] I tried *8 and it did not work. Can a IAXs client also me assigned into a call-pickup group? Thanks in advance, Stephen ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Pickup issue
I know this has been discussed heavily but i have a bizzare issue with call pickup. I have 3 asterisk servers all built the same on centos 4.1 and call pickup works on two of them but not on the third. They have identical configurations. I'm using asterisk 1-0-9 and zaptel with ztdummy. All phones are sip with a mix of sip clients and cisco 7960's. All i get in the asterisk debug is - Sep 22 09:32:50 NOTICE[9562]: chan_sip.c:7427 handle_request: Nothing to pick up Anyone offer any ideas? Ta! ___ Yahoo! Messenger - NEW crystal clear PC to PC calling worldwide with voicemail http://uk.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Pickup between ZAP and SIP technologies
Hi, I have this scenario. In my desk I have a phone connected to a FXS module of my * server. On another desk there is a phone but it is a SIP softphone (SJphone). I hear the SIP softphone is ringing, then I try to take that call with my Zap phone in my desk dialing *8, but I get fast busy tone. Is there I way do this to work ? I mean pickup phones that are ringing on different technologies ? Ardg. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Pickup with Dialog on snom display
Hello Everybody, 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? Regards Bastian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call pickup with a variable pickupgroup/callergroup based on context
Hi If all my calls are hitting one context in sip.conf, from where they are passed to extensions.conf, and then passed to appropriate contexts, how can I have pickupo features enabled per context, and also define pickupgroups, or can i use a variable in sip.conf for this. I had a lookup at the bristuff setup, and this seems to do it, but I was hoping for a simpler :-) setup Iqbal ___ 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
AW: [Asterisk-Users] call pickup with snom function keys now working with cvs-head + patch sipsubscribe-20050715.rev779.txt
> hi listmembers, > > please test my new patch to chan_sip.c which is to make call > pickup on > the snom phones (and maybe other phones that support > 'INVITE/Replaces') > work and make comments in the bugtracker > http://bugs.digium.com/view.php?id=3644 so it can make its > way into the cvs. This really sounds very exciting. Excuse my beginner´s question: Where and how do I get this patch and how do I install a patch like this in an environment that should not be stopped for more than a few minutes? Is it any easy step to install this patch? Best regards, Armin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call pickup with snom function keys now working with cvs-head + patch sipsubscribe-20050715.rev779.txt
hi listmembers, please test my new patch to chan_sip.c which is to make call pickup on the snom phones (and maybe other phones that support 'INVITE/Replaces') work and make comments in the bugtracker http://bugs.digium.com/view.php?id=3644 so it can make its way into the cvs. this patch sipsubscribe-20050715.rev779.txt enables: * monitoring of other lines (using the 'hint' priority) - LED off when monitored phone idle - LED on when monitored phone busy - LED blinking when monitored phone ringing * display of caller id on monitoring phone * call pickup by pressing function key beneath the blinking led * corrected MWI LED functionality * corrected MWI button functionality * dialplan extension for MWI button settable * major code cleanup and some other things i don't remember. with this patch the snom phone will _THE_ phone for a receptionist - and for every phone usergroup that likes easy call pickup. please test it extensivly and comment it in the bugtracker as i think many of us have been long waiting for this functionality (and it's a real pain for me to keep it up to date as it is a real big patch) regards frank (xylome) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call pickup with snom phones
hi, is there anybody who was able to setup call pickup with a snom phone? searching through the web brought up this: http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+snom section "call pickup" but this doesen't seem to work with current releases of the snom firmware (and looking through the patch of easywe it never worked very good at all) current snom firmware doesn't seem to send the required INVITE/REPLACE messages. any help is appreciated. regards frank ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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
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 -- #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] call pickup with Sipura-3000
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? -- #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 " 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 "...
Philipp
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[Asterisk-Users] Call pickup across technologies (SIP, IAX, MGCP)?
Hi there, 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 1: SIP phone 1 ringing - SIP phone 2 can pick the call up with *8 We are happy! :-) * Case 2: IAX phone ringing - SIP phone can't pick the call up: NOTICE[10250]: Nothing to pick up * Case 3: SIP phone ringing - IAX phone can't pick the call up: NOTICE[12300]: Rejected connect attempt from 192.168.x.y, reque st '[EMAIL PROTECTED]' does not exist The same applies to MGCP and SIP phone interaction. [features.conf] pickupexten = *8 [sip.conf and iax.conf] callgroup=1 pickupgroup=2 Cheers, 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
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
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
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. -- 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
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
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
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
[Asterisk-Users] Call Pickup
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] Call Pickup Problem
I'm having a problem with the call pickup with the latest CVS. Before I updated to the latest CVS it was working fine. Now, whenever anyone tries to pickup a call using *8 it dumps all calls going on at the time and hangs up on the incoming call. Tim ___ 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 - 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
- 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
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 Danışman - Technical
Consultant
Grid Bilişim
Teknolojileri A.Ş.
Kuştepe Mahallesi Leylak
Sokak
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Daire:9
34387 Şişli İstanbul
Türkiye
Tel :
+90 (212) 336 92 55
Fax : +90
(212) 266 25 50
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
LeandroSent: 19 Kasım 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:
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
> > 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 callgroup=2 channel => 1 In sip.conf, an phone entry might look like: [3002] type= username=3002 secret= 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 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
- 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 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
[Asterisk-Users] Call pickup
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? 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
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
[Asterisk-Users] Call Pickup
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. Jerry ___ 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 and snom phones
You need to have the pickupgroups added in sip.conf Then - in order to pick up, use *8 ( and not *8# ). Under each extension ( here in group 1 ) add the following lines to sip.conf : callgroup=1 pickupgroup=1 -- Pertti [EMAIL PROTECTED] wrote: First of all, excuse me if this is considered as OT. I'm trying to use the asterisk call pickup function on the 220 Snom phones, in other phones works well. But if I dial *8# in the snom phones, the call is no picked up. In others phones this combo of keys works perfectly. Someone could give me a clue? Any info will be appreciated. Ismael ___ 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
[Asterisk-Users] Call pickup and snom phones
First of all, excuse me if this is considered as OT. I'm trying to use the asterisk call pickup function on the 220 Snom phones, in other phones works well. But if I dial *8# in the snom phones, the call is no picked up. In others phones this combo of keys works perfectly. Someone could give me a clue? Any info will be appreciated. Ismael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call pickup (group)
Hi, Asterisk has a feature called pickupgroup, meaning you can pickup the call that is ringing on your collegues phone. Can this type of behaviour be emulated in extension logic or AGI (maybe together with manager login) ? We need the group settings to be tied into a database which makes it a little more dynamic :-> Any suggestions are welcome. Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Pickup problem in Asterisk with SIP phones
I'm having a tough time getting call pickup to work on *. Here's my
configuration:
X100P with T-1, channels 1-4 voice <---> * <---CISCO 7960 with SIP 6.0 Image
A call comes in, and * picks up and presents a menu. Caller chooses
extension, (in this case ext 103, SIP/wsmith)
Wsmith is sitting in my office, hears his phone ringing, picks up my phone,
gets dial tone, and presses *8. He gets a reorder (fast busy) on my phone,
and his phone continues to ring (he then curses loudly, and goes racing down
the hall to try to catch the call)
In * , I get a
Jun 9 15:45:14 NOTICE[229391]: chan_sip.c:5417 handle_request: Nothing to
pick up
I turned on SIP debugging, cleaned out all the Sip register messages that
were flying about while debugging, and present the logs here. My version is
CVS-05/24/04
My zapata.conf looks like:
group=1
callgroup=1
pickupgroup=1-4
context=NuFone-Outgoing
signalling = fxs_ks
callprogress=no
callerid="Radiance Technologies" <(251)-445-0045>
usecallerid=yes
My SIP.conf looks like:
sip.conf[] 0 L:[105+37 142/142] *(3505/3516b)= c 99 0x63
dtmfmode=inband
mailbox=102
context=Outgoing
callerid="Dean Li" <102>
username=dli
secret=rad1ance
pickupgroup=1
;the ringing SIP phone:
[wsmith]
type=friend
host=dynamic
nat=yes
canreinvite=no
qualify=1000
;defaultip=192.168.30.108
dtmfmode=inband
mailbox=103
context=Outgoing
callerid="Walter Smith" <103>
username=wsmith
secret=**
pickupgroup=1-4
;The phone attempting the *8
[nmartin]
type=friend
host=dynamic
insecure=no
nat=yes
canreinvite=no
qualify=1000
;defaultip=192.168.30.100
dtmfmode=inband
mailbox=105
context=Outgoing
callerid="Nik Martin" <105>
username=nmartin
secret=**
pickupgroup=1-4
callgroup=1
The SIP debug:
pbxMobile*CLI>
-- Starting simple switch on 'Zap/1-1'
pbxMobile*CLI>
-- Executing Wait("Zap/1-1", "3") in new stack
pbxMobile*CLI>
-- Executing Answer("Zap/1-1", "") in new stack
pbxMobile*CLI>
-- Executing NoOp("Zap/1-1", ""MOBILE, AL" ") in new stack
pbxMobile*CLI>
-- Executing Wait("Zap/1-1", "1") in new stack
pbxMobile*CLI>
Jun 9 15:45:02 WARNING[2211866]: chan_zap.c:3073 zt_handle_event:
Ring/Off-hook in strange state 6 on channel 1
pbxMobile*CLI>
-- Executing BackGround("Zap/1-1", "radiancewelcome") in new stack
pbxMobile*CLI>
-- Playing 'radiancewelcome' (language 'en')
pbxMobile*CLI>
11 headers, 2 lines
8 headers, 0 lines
pbxMobile*CLI>
== CDR updated on Zap/1-1
pbxMobile*CLI>
-- Executing Dial("Zap/1-1", "SIP/wsmith|20|tT") in new stack
pbxMobile*CLI>
We're at 172.31.30.3 port 15418
pbxMobile*CLI>
Answering with preferred capability 4
pbxMobile*CLI>
Answering with preferred capability 2
pbxMobile*CLI>
12 headers, 9 lines
pbxMobile*CLI>
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP
172.31.30.3:5060;branch=z9hG4bK5cc08566 From: "MOBILE, AL"
;tag=as05f4b37a To:
Contact: Call-ID:
[EMAIL PROTECTED] CSeq: 102 INVITE User-Agent:
Asterisk PBX Date: Wed, 09 Jun 2004 20:45:09 GMT Allow: INVITE, ACK, CANCEL,
OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 181 v=0
o=root 20260 20260 IN IP4 172.31.30.3 s=session c=IN IP4 172.31.30.3 t=0 0
m=audio 15418 RTP/AVP 0 3 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000
a=silenceSupp:off - - - - (NAT) to 172.31.30.11:5060
pbxMobile*CLI>
-- Called wsmith
pbxMobile*CLI>
Sip read:
SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.31.30.3:5060;branch=z9hG4bK5cc08566
From: "MOBILE, AL" ;tag=as05f4b37a To:
Call-ID:
[EMAIL PROTECTED] Date: Wed, 09 Jun 2004 20:48:00
GMT CSeq: 102 INVITE Server: CSCO/6 Contact:
Content-Length: 0
10 headers, 0 lines
pbxMobile*CLI>
Sip read:
SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.31.30.3:5060;branch=z9hG4bK5cc08566
From: "MOBILE, AL" ;tag=as05f4b37a To:
;tag=000b46e9ae7e485f2abff4bc-43940b23 Call-ID:
[EMAIL PROTECTED] Date: Wed, 09 Jun 2004 20:48:00
GMT CSeq: 102 INVITE Server: CSCO/6 Contact:
Content-Length: 0
10 headers, 0 lines
pbxMobile*CLI>
-- SIP/wsmith-7e27 is ringing
pbxMobile*CLI>
pbxMobile*CLI>
Sip read:
INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP
172.31.30.7:5060;branch=z9hG4bK2408959c From: "105 - Nik Martin"
;tag=003094c4481f49565aff56ad-226e8953 To:
Call-ID:
[EMAIL PROTECTED] Date: Wed, 09 Jun 2004
20:47:30 GMT CSeq: 101 INVITE User-Agent: CSCO/6 Contact:
Expires: 180 Content-Type: application/sdp
Content-Length: 244 Accept: application/sdp Remote-Party-ID: "105 - Nik
Martin"
;party=calling;id-type=subscriber;privacy=off;scree
n=no v=0 o=Cisco-SIPUA 24482 2915 IN IP4 172.31.30.7 s=SIP Call c=IN IP4
172.31.30.7 t=0 0 m=audio 26676 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
14 headers, 11 lines
Using latest request as basis request
Sending to 172.31.30.7 : 5060 (non-NAT)
Found audio format UNKN
Found audio format ALAW
Found audio format UNKN
Found audio format UNKN
Found descripti
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 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
[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.
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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
[Asterisk-Users] call pickup fails.
Hello all. 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 ? Thank's. ___ 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 - 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
[Asterisk-Users] Call pickup - still keeps ringing?
Title: Call pickup - still keeps ringing? On my system with 0.7.1 call pickup from SIP to SIP still leaves the originally dialed phone ringing for 10's of seconds after the call has been picked up on another line. There was a post a back in the fall that said this had been broken in a code update. Does 0.7.1 still have this problem or is it my user error? To be clear, I can do the call pickup with *8 just fine. However, the phone that was called keeps ringing for about 30 seconds ~after~ I've done the pickup on another phone. I can shorten this by shortening the response timeout but that doesn't work for the customer. Thanks.
