Re: [asterisk-users] call forwarding number from outside.
Hello, I have a similar problem. Whenever a call comes in to my asterisk I handle it like this: exten = s,1, Answer() exten = s, n, Dial(SIP/exten,20,fotT) exten = s, 1, Hangup() it works fine but in the SIP messages th IP-Address from Asterisk is in the From field. For example I am calling from 123@134.32.220.33 then the SIP message behind the Asterisk looks like INVITE sip:2232@10.10.10.11 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.10 (Asterisk IP) From: 123 sip:soft@10.10.10.10 (again Asterisk IP) how can I change this? Thanks Tom -- sip:3...@perenaster.com sip:3...@perenaster.com sip:3...@perenaster.com sip:3...@perenaster.com sip:3...@perenaster.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as a Operator Phone
On Tue, Jul 26, 2011 at 12:37 AM, Nikhil d.nik...@cem-solutions.net wrote: I am using asterisk as a client not as a server. For client I need features like transfer ,call forward ,multiple lines as in normal IP Phones like CISOC,polycom. In asterisk ,we have chan_alsa driver that will communicate to the local soundcard. If I installed asterisk in my ubuntu system,and using CLI command I can make calls outside and once call connected I can hear and talk from my Headphone. I planing to enhance chan_alsa module to get the features same as in SIP client. Thanks Nikhil On 07/26/2011 12:57 AM, Duncan Turnbull wrote: Asterisk can run operator phones with no problem, there are multiple phones out there with addon buttons for automating shared line appearances forwards and other functions For example yealink have the t38 with 6 lines and 16 buttons and the ex 38 with 38 additional programmable buttons to add to that if you need Are you talking about a phone that is not sip based? I am not sure why you need to use chan_alsa? Cheers Duncan Sent from my iPhone please excuse the typos On 25/07/2011, at 12:30 AM, Nikhild.nikhil@cem-solutions.**netd.nik...@cem-solutions.net wrote: Any reply on this.. On 07/22/2011 12:56 PM, Nikhil wrote: Hi Does anyone used asterisk as a operator phone,with multiple lines and features like transfer forward and etc.I used chan_alsa driver to make asterisk as SIP Phone,but it has limitation,we cant make or receive multiple calls,and will not able to do any features like transfer forward etc. Is any other application available in asterisk to do this . Thanks Nikhil -- __**__** _ -- 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-**usershttp://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-**usershttp://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-**usershttp://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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users Using asterisk as a client sounds interesting. I guess all the existing sip clients suck? -Kyle -- _ -- 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] asterisk + sccp-b problem
Dear, with asterisk 1.6.2.18 and sccp-bv3stable on two servers, we tried to register about 1200 cisco phones, for a company. in out of official hours, all 1200 phones registered and the cpu and ram was below 5%. H323 is the protocol for incoming calls, and SIP for outgoing ones. in official hours, with only 10 calls, the cpu went more than 100% , and crashed. the bt full result of gdb was attached I have some questions now, 1-is any problem in the attached report. 2-does asterisk 1.4 more stable than 1.6 in this case? -- Pezhman Lali --gdb bt full--- #0 ast_rtp_set_peer (rtp=0x0, them=0x59d1b450) at rtp.c:2707 No locals. #1 0x2aaac9a4c153 in setup_rtp_connection (call_reference=32275, remoteIp=0x2aaadc0d6a40 10.11.1.11, remotePort=value optimized out, token=0x4d01c60 ip$10.11.1.11:48440/32275, pt=0) at chan_h323.c:1975 pvt = 0x4cd13e0 them = {sin_family = 2, sin_port = 1096, sin_addr = {s_addr = 184617738}, sin_zero = \240\264\037��*\000} rtp_change = value optimized out __PRETTY_FUNCTION__ = setup_rtp_connection #2 0x2aaac9a5cac9 in MyH323_ExternalRTPChannel::Start() () from /usr/lib/asterisk/modules/chan_h323.so No symbol table info available. #3 0x003dd34dc237 in H323Connection::OnSelectLogicalChannels() () from /usr/lib/libh323_linux_x86_64_r.so.1.18.0 No symbol table info available. #4 0x003dd34d4294 in H323Connection::SendFastStartAcknowledge(H225_ArrayOf_PASN_OctetString) () from /usr/lib/libh323_linux_x86_64_r.so.1.18.0 No symbol table info available. #5 0x003dd34d938e in H323Connection::AnsweringCall(H323Connection::AnswerCallResponse) () from /usr/lib/libh323_linux_x86_64_r.so.1.18.0 No symbol table info available. #6 0x003dd34db448 in H323Connection::OnReceivedSignalSetup(H323SignalPDU const) () from /usr/lib/libh323_linux_x86_64_r.so.1.18.0 No symbol table info available. #7 0x2aaac9a5e8cd in MyH323Connection::OnReceivedSignalSetup(H323SignalPDU const) () from /usr/lib/asterisk/modules/chan_h323.so No symbol table info available. #8 0x003dd34db8c8 in H323Connection::HandleSignalPDU(H323SignalPDU) () from /usr/lib/libh323_linux_x86_64_r.so.1.18.0 No symbol table info available. #9 0x2aaac9a5d5dd in MyH323Connection::HandleSignalPDU(H323SignalPDU) () from /usr/lib/asterisk/modules/chan_h323.so No symbol table info available. #10 0x003dd3514808 in H323Transport::HandleFirstSignallingChannelPDU() () from /usr/lib/libh323_linux_x86_64_r.so.1.18.0 No symbol table info available. #11 0x003dd3514940 in H225TransportThread::Main() () from /usr/lib/libh323_linux_x86_64_r.so.1.18.0 No symbol table info available. #12 0x003552c837ac in PThread::PX_ThreadStart(void*) () from /usr/lib64/libpt_linux_x86_64_r.so.1.10.3 No symbol table info available. #13 0x003c5860673d in start_thread () from /lib64/libpthread.so.0 No symbol table info available. #14 0x003c57ed40cd in clone () from /lib64/libc.so.6 No symbol table info available. -- _ -- 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