Re: [asterisk-users] Re: calls to SPA942 disconnect after 15 seconds (chan_sip.c set_destination: can't find address)
Thanks, I'll give it a try. Allan. - Original Message From: M.Hockings [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, January 11, 2007 3:45:44 PM Subject: [asterisk-users] Re: calls to SPA942 disconnect after 15 seconds (chan_sip.c set_destination: can't find address) Allan Kamau wrote: Am having a unique problem, calls received on my SPA942 seem to end after 15 seconds, but calls made from this device do not have this problem. For this device (when receiving calls) I get periodic chan_sip.c set_destination: can't find address for host I have set the canreinvite=no in the sip.conf. Does anyone have a sample entry from sip.conf for the Lynksys SPA 942 to share with me. Allan. Hi Allan, This is probably singularly unspectacular but is characteristic of what I have for SPA922 (single line version of the 942). Mike username=3200 type=friend secret= record_out=On-Demand record_in=On-Demand qualify=yes port=5060 nat=never [EMAIL PROTECTED], [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=3200 3200 ___ --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 Have a burning question? Go to www.Answers.yahoo.com and get answers from real people who know. ___ --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] calls to SPA942 disconnect after 15 seconds (chan_sip.c set_destination: can't find address)
Am having a unique problem, calls received on my SPA942 seem to end after 15 seconds, but calls made from this device do not have this problem. For this device (when receiving calls) I get periodic chan_sip.c set_destination: can't find address for host I have set the canreinvite=no in the sip.conf. Does anyone have a sample entry from sip.conf for the Lynksys SPA 942 to share with me. Allan. Yahoo! Music Unlimited Access over 1 million songs. http://music.yahoo.com/unlimited ___ --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] Invite issues
Hi all, I am receiving handle_request_invite: Failed to authenticate errors on two VoIP gateway devices connected to my asterisks SIP server. The problem seems to be in my configuration. I will only focus on one of this devices in this mail. On this device I receive the error Sep 21 12:02:23 NOTICE[30782]: chan_sip.c:10468 handle_request_invite: Failed to authenticate user 2006 sip:[EMAIL PROTECTED]:5060;tag=43c7d6ab at the CIL This is the configuration for 2006 in my sip.conf [2006] type=friend username=2006 secret=2006 context=from-sip callerid=Allan 2006 host=dynamic defaultip=192.168.0.100 ;nat=no canreinvite=yes dtmfmode=RFC2833 [12] insecure=very canreinvite=yes type=friend username=12 secret=12 context=from-gsm callerid=Allan 12 host=dynamic ;defaultip=192.168.0.100 dtmfmode=rfc2833 ;register=:@192.168.0.10 ; Local interface ;qualify=no Attached kindly find the SIP communication captured between the device and Asterisk. Allan. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com No. TimeSourceDestination Protocol Info 8 7.567552192.168.0.100 192.168.0.2 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description Frame 8 (860 bytes on wire, 860 bytes captured) Arrival Time: Sep 21, 2006 09:39:48.143706000 Time delta from previous packet: 4.270322000 seconds Time since reference or first frame: 7.567552000 seconds Frame Number: 8 Packet Length: 860 bytes Capture Length: 860 bytes Protocols in frame: eth:ip:udp:sip:sdp Coloring Rule Name: UDP Coloring Rule String: udp Ethernet II, Src: PortechC_00:01:8c (00:03:7e:00:01:8c), Dst: Zioncom_f1:de:eb (00:0e:e8:f1:de:eb) Destination: Zioncom_f1:de:eb (00:0e:e8:f1:de:eb) Address: Zioncom_f1:de:eb (00:0e:e8:f1:de:eb) ...0 = Multicast: This is a UNICAST frame ..0. = Locally Administrated Address: This is a FACTORY DEFAULT address Source: PortechC_00:01:8c (00:03:7e:00:01:8c) Address: PortechC_00:01:8c (00:03:7e:00:01:8c) ...0 = Multicast: This is a UNICAST frame ..0. = Locally Administrated Address: This is a FACTORY DEFAULT address Type: IP (0x0800) Internet Protocol, Src: 192.168.0.100 (192.168.0.100), Dst: 192.168.0.2 (192.168.0.2) Version: 4 Header length: 20 bytes Differentiated Services Field: 0xa0 (DSCP 0x28: Class Selector 5; ECN: 0x00) 1010 00.. = Differentiated Services Codepoint: Class Selector 5 (0x28) ..0. = ECN-Capable Transport (ECT): 0 ...0 = ECN-CE: 0 Total Length: 846 Identification: 0x4111 (16657) Flags: 0x00 0... = Reserved bit: Not set .0.. = Don't fragment: Not set ..0. = More fragments: Not set Fragment offset: 0 Time to live: 61 Protocol: UDP (0x11) Header checksum: 0xb737 [correct] Good: True Bad : False Source: 192.168.0.100 (192.168.0.100) Destination: 192.168.0.2 (192.168.0.2) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Source port: 5060 (5060) Destination port: 5060 (5060) Length: 826 Checksum: 0x2331 [correct] Session Initiation Protocol Request-Line: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Method: INVITE Resent Packet: False Message Header Via: SIP/2.0/UDP 192.168.0.100:5060;rport;branch=z9hG4bK63d97bbf1956f9ad23a40b1dcb898fcc From: 2006 sip:[EMAIL PROTECTED]:5060;tag=74f33ef1 SIP Display info: 2006 SIP from address: sip:[EMAIL PROTECTED]:5060 SIP tag: 74f33ef1 To: sip:[EMAIL PROTECTED] SIP to address: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]:5060 Contact Binding: sip:[EMAIL PROTECTED]:5060 URI: sip:[EMAIL PROTECTED]:5060 SIP contact address: sip:[EMAIL PROTECTED]:5060 CSeq: 801 INVITE Max-Forwards: 70 Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE Content-Type: application/sdp User-Agent: CMI CM5K Content-Length: 329 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 56901 0 IN IP4 192.168.0.100 Owner Username: - Session ID: 56901 Session Version: 0 Owner Network Type: IN Owner Address Type: IP4 Owner Address: 192.168.0.100 Session Name (s): SIP CALL Connection Information (c): IN IP4 192.168.0.100 Connection Network Type: IN Connection Address Type: IP4
[asterisk-users] SPA-3102 PSTN-VoIP Gateway (quest for one stage dialing)
I have read in the manual the SPA-3102 does not support one stage dialing for PSTN-VoIP calls, this is indeed frustrating, is there a way to provide seamless PSTN-VoIP gateway calls. I finally managed to get the SPA-3102 to work this is done by connecting via the ethernet port named Internet, I can now call via the PSTN. Allan. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SPA 3102 does not even attempt to register
I don't see a -- Saved useragent line for the SPA 3102 device am trying to connect to Asterisk. I have similar configuration for the SPA 3102 as I have for another hard phone in the sip.conf file but the device (SPA 3102) does not even attempt to register. I have configured the device to register with the sip proxy 192.168.0.2 but nothing happens. Using sip show peers at the CLI this is what I see. sip show peers Name/username HostDyn Nat ACL Port Status 2001/2001 192.168.0.11 D 5060 Unmonitored 2006/2006 192.168.0.100D 5060 Unmonitored myAGI-app/myagi_app192.168.0.10 5060 Unmonitored 2008/2008 (Unspecified)D 0Unmonitored 4 sip peers [4 online , 0 offline] __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] inbound call from GSM gateway: handle_request_invite: Failed to authenticate user
Hi all, I am getting a handle_request_invite: Failed to authenticate user error when I attempt to receive calls from a GSM gateway (I can successfully call through the device VoIP-GSM from asterisk). I have looked for a solution to this error but most point me to adding a register line which I've tried (or probably my syntax may be wrong) without success, below is an extract from the CLI, below that is my sip.conf CLI Use EXIT or QUIT to exit the asterisk console -- Registered SIP '2006' at 192.168.0.100 port 5060 expires 60 -- Saved useragent CMI CM5K for peer 2006 Sep 15 18:11:02 NOTICE[19600]: chan_sip.c:10468 handle_request_invite: Failed to authenticate user +27729932161 sip:[EMAIL PROTECTED]:5060;tag=3ed9e889 /CLI [general] context=from-sip; Default context for incoming calls ; if asterisk was compiled with OSP support. ; defaults to asterisk ; Realms MUST be globally unique according to RFC 3261 ; Set this to your host name or domain name bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) insecure=very srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;register=2006:[EMAIL PROTECTED]:5060 [authentication] [2008] type=friend username=2008 secret=XXX context=from-sip; Where to start in the dialplan when this phone calls host=dynamic defaultip=192.168.0.12 port=5061 insecure=very canreinvite=yes [aaron] type=friend username=aaron secret=XXX context=from-sip; Where to start in the dialplan when this phone calls host=dynamic defaultip=192.168.0.12 insecure=very canreinvite=yes [myAGI-app] type=friend username=myagi_app secret=XXX context=from-sip; Where to start in the dialplan when this phone calls callerid=XXX host=XXX [2006] insecure=very type=friend username=2006 secret=XXX context=from-gsm; Where to start in the dialplan when this phone calls callerid=XXX host=dynamic defaultip=192.168.0.100 dtmfmode=rfc2833 [2001] type=friend username=2001 secret=XXX context=from-sip; Where to start in the dialplan when this phone calls callerid=XXX host=dynamic defaultip=192.168.0.11 ; No registration allowed dtmfmode=RFC2833; either RFC2833 or INFO for the BudgeTone __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] running agi application in the background
I would like to run a fast-agi application in the background.(cmd agi()) This is because I would like to implement a disconnect after so many seconds feature or at least a log of the duration of the call. When the call is answered, the application checks to see the number of seconds (talk time)remaining then disconnects the call if the time is exceeded. How can I achieve this. Allan. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dialing out using SIP terminal
Hi all,
I have a VoIP GSM (SIP) terminal that I have
successfully configured and registered in asterisk,
would like to:
a)Answer calls via asterisk coming from this terminal.
b)Route outbound calls to this terminal.
What Dial command do I use so as to have the sip
terminal dial an outside line, for example when using
Zap, I use the following commands successfully.
extension.conf
[from-sip]
exten=20,1,Dial(Zap/4/0w10136)
exten=_22.,1,Dial(Zap/4/0w${EXTEN:2},5,r)
If I use the command below in an attempt to dial a
number 0729932165 on the SIP/2006 channel, the call is
answered by the terminal but doesn't dial out.
exten=21,1,Dial(SIP/2006/0729932165,5,r)
My current configuration looks like this
sip.conf
[2006]
type=friend
username=2006
secret=2006
context=voip_gsm; Where to start in the dialplan
when this phone calls
callerid=Allan 2006 ; Full caller ID, to override
the phones config
host=dynamic
defaultip=192.168.0.100
extension.conf
[voip_gsm]
exten=s,1,NoOp(${EXTEN})
exten=s,2,NoOp(${CALLERID})
exten=s,3,Dial(SIP/2001,5,Ttm) ;dial a sip hardphone
configured in my Asterisk installation.
exten=s,4,Voicemail([EMAIL PROTECTED])
exten=s,5,Hangup
exten=s,104,Voicemail([EMAIL PROTECTED])
exten=s,105,Hangup
[from-sip]
exten=21,1,Dial(SIP/2006/0729932165,5,r)
Allan.
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Re: [Asterisk-Users] GSM gateway hardware
2 to 4 channels to start with. Allan. --- hakem voip [EMAIL PROTECTED] wrote: How many channels do you need per gateway ? I might have slution for you voip2gsm Regards On 7/20/05, Allan Kamau [EMAIL PROTECTED] wrote: Thanks Roger, I find the second option more interesting, let me know once you've managed to provide asterisk support for the GSM modem. Allan. --- Roger Schreiter [EMAIL PROTECTED] wrote: Allan Kamau schrieb: ... I am looking for a GSM VoIP gateway for use with Hi, do you think of something to interconnect to GSM carriers via cable (GSM-A) or do you think about using a GSM-modem with all its limitations? For the first option I could forward your email address to someone providing GSM-A stacks for asterisk. For the second option, it might be interesting for you, that we are currently also working on asterisk support for a GSM-modem. 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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] GSM gateway hardware
Hi All, I am looking for a GSM VoIP gateway for use with Asterisk. I have come across VoiceBlue by 2N but it's price is beyond my reach. Are there any other alternatives out there? I've scanned across the mail achieves for an answer to this without much success, if the question has already been answered kindly point me to the resource. Allan. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM gateway hardware
Thanks Roger, I find the second option more interesting, let me know once you've managed to provide asterisk support for the GSM modem. Allan. --- Roger Schreiter [EMAIL PROTECTED] wrote: Allan Kamau schrieb: ... I am looking for a GSM VoIP gateway for use with Hi, do you think of something to interconnect to GSM carriers via cable (GSM-A) or do you think about using a GSM-modem with all its limitations? For the first option I could forward your email address to someone providing GSM-A stacks for asterisk. For the second option, it might be interesting for you, that we are currently also working on asterisk support for a GSM-modem. 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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
