Re: [asterisk-users] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used
On Wed, Aug 4, 2010 at 10:25 PM, Joe Wood wrote: > On Wed, Aug 4, 2010 at 7:49 PM, Warren Selby > wrote: > > On Wed, Aug 4, 2010 at 9:25 PM, Joe Wood wrote: > >> > > My experience with Asterisk in the past has been with inbound analog > lines so that would make sense :) > > See if you spot anything weird here: > > Try adding "insecure=invite" to the DID_NUMBER peer, reload SIP and try your call again. By the way, it looks like your SIP provider has a built-in auto-failover to voicemail setup. You may want to get them to disable that once you get everything working on your end. -- Thanks, --Warren Selby http://www.selbytech.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] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used
On Wed, Aug 4, 2010 at 7:49 PM, Warren Selby wrote: > On Wed, Aug 4, 2010 at 9:25 PM, Joe Wood wrote: >> >> I don't see any >> >> On Wed, Aug 4, 2010 at 7:04 PM, Warren Selby >> wrote: >> > >> > You don't have any extensions in your default context that match the >> > extension that your sip peer is dialing in on. 's' is not a default >> > extension for SIP...try using _X., and see what you get. Bump up the >> > CLI >> > (core set verbose 10) and then repost a failed called attempt. Some SIP >> > providers also use a + symbol in front of their inbound calls, so you >> > may >> > need to use _+X., instead. >> >> I don't see any call attempt/logs when I bump up the verbosity, and >> when I check my verbose logs I show: >> > > The next step would be to enable sip debug on the peer you're trying to > receive calls from (sip set debug peer PEERNAME or sip set debug ip > IPADDRESS). Then send another call inbound and see what's happening. As > far as the 's' extension, that's the "start" extension, it's used when no > other extension information is presented. Pretty much every SIP peer I've > ever seen presents an extension when entering a context, and thus the 's' > extension doesn't catch it. I've typically only seen 's' used in Macros and > with inbound analog lines. > My experience with Asterisk in the past has been with inbound analog lines so that would make sense :) See if you spot anything weird here: <--- SIP read from UDP:209.221.186.98:5060 ---> INVITE sip:s...@209.221.186.50 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 209.221.186.98;branch=z9hG4bK6b95.ae84c9620e19761029084a0f85255c29.0 Via: SIP/2.0/UDP 209.221.186.98:5071;branch=z9hG4bKf0a046579186bf058ba7783ed98a5a66;rport=5071 Max-Forwards: 16 From: 2538544199 ;tag=f7093e2d7e16a927d0816f6f5ed7aba4 To: Call-ID: ba711218-1ae3-122e-41bd-18a905721e58-b2b_1 CSeq: 200 INVITE Contact: Anonymous Expires: 300 User-Agent: Sippy Softswitch v2.0.80 cisco-GUID: 1225641884-3786690633-966044271-4144140181 h323-conf-id: 1225641884-3786690633-966044271-4144140181 Content-Length: 321 Content-Type: application/sdp v=0 o=- 1280279699622 1280279699622 IN IP4 209.221.186.98 s=- c=IN IP4 209.221.186.98 t=0 0 m=audio 60304 RTP/AVP 0 a=fmtp:4 bitrate=6300;annexa=no a=rtpmap:96 iLBC/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=fmtp:18 annexb=no a=rtpmap:98 telephone-event/8000 a=oldmediaip:208.76.155.20 a=nortpproxy:yes <-> [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 0 [ 35]: INVITE sip:s...@209.221.186.50 SIP/2.0 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 1 [ 75]: Record-Route: [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 2 [ 85]: Via: SIP/2.0/UDP 209.221.186.98;branch=z9hG4bK6b95.ae84c9620e19761029084a0f85255c29.0 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 3 [ 94]: Via: SIP/2.0/UDP 209.221.186.98:5071;branch=z9hG4bKf0a046579186bf058ba7783ed98a5a66;rport=5071 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 4 [ 16]: Max-Forwards: 16 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 5 [ 85]: From: 2538544199 ;tag=f7093e2d7e16a927d0816f6f5ed7aba4 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 6 [ 35]: To: [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 7 [ 51]: Call-ID: ba711218-1ae3-122e-41bd-18a905721e58-b2b_1 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 8 [ 16]: CSeq: 200 INVITE [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 9 [ 55]: Contact: Anonymous [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 10 [ 12]: Expires: 300 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 11 [ 36]: User-Agent: Sippy Softswitch v2.0.80 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 12 [ 54]: cisco-GUID: 1225641884-3786690633-966044271-4144140181 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 13 [ 56]: h323-conf-id: 1225641884-3786690633-966044271-4144140181 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 14 [ 19]: Content-Length: 321 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 15 [ 29]: Content-Type: application/sdp [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 16 [ 0]: [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request:Body 0 [ 3]: v=0 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request:Body 1 [ 53]: o=- 1280279699622 1280279699622 IN IP4 209.221.186.98 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request:Body 2 [ 3]: s=- [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request:Body 3 [ 23]: c=IN IP4 209.221.186.98 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request:Body 4 [ 5]: t=0 0 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request:Body 5 [ 23]: m=aud
Re: [asterisk-users] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used
On Wed, Aug 4, 2010 at 9:25 PM, Joe Wood wrote: > I don't see any > > On Wed, Aug 4, 2010 at 7:04 PM, Warren Selby > wrote: > > > > You don't have any extensions in your default context that match the > > extension that your sip peer is dialing in on. 's' is not a default > > extension for SIP...try using _X., and see what you get. Bump up the CLI > > (core set verbose 10) and then repost a failed called attempt. Some SIP > > providers also use a + symbol in front of their inbound calls, so you may > > need to use _+X., instead. > > I don't see any call attempt/logs when I bump up the verbosity, and > when I check my verbose logs I show: > > The next step would be to enable sip debug on the peer you're trying to receive calls from (sip set debug peer PEERNAME or sip set debug ip IPADDRESS). Then send another call inbound and see what's happening. As far as the 's' extension, that's the "start" extension, it's used when no other extension information is presented. Pretty much every SIP peer I've ever seen presents an extension when entering a context, and thus the 's' extension doesn't catch it. I've typically only seen 's' used in Macros and with inbound analog lines. -- Thanks, --Warren Selby http://www.selbytech.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] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used
I don't see any On Wed, Aug 4, 2010 at 7:04 PM, Warren Selby wrote: > > You don't have any extensions in your default context that match the > extension that your sip peer is dialing in on. 's' is not a default > extension for SIP...try using _X., and see what you get. Bump up the CLI > (core set verbose 10) and then repost a failed called attempt. Some SIP > providers also use a + symbol in front of their inbound calls, so you may > need to use _+X., instead. I don't see any call attempt/logs when I bump up the verbosity, and when I check my verbose logs I show: [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Registered extension context 'default' (0xb77980c0) in local table 0xb77960c0; registrar: pbx_config [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.' priority 1 to default (0xb77980c0) [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.' priority 2 to default (0xb77980c0) [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.' priority 3 to default (0xb77980c0) [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.' priority 4 to default (0xb77980c0) [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.' priority 5 to default (0xb77980c0) [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.' priority 6 to default (0xb77980c0) [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.' priority 7 to default (0xb77980c0) [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.' priority 8 to default (0xb77980c0) [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Registered extension context 'parkedcalls' (0xb7797ee0) in local table 0xb77960c0; registrar: features [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- merging incls/swits/igpats from old(parkedcalls) to new(parkedcalls) context, registrar = pbx_config [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '700' priority 1 to parkedcalls (0xb7797ee0) [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Time to scan old dialplan and merge leftovers back into the new: 0.89 sec [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Time to restore hints and swap in new dialplan: 0.02 sec [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Time to delete the old dialplan: 0.11 sec [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Total time merge_contexts_delete: 0.000102 sec [Aug 4 19:17:04] VERBOSE[12255] netsock.c: == Using SIP RTP CoS mark 5 [Aug 4 19:19:04] VERBOSE[12255] netsock.c: == Using SIP RTP CoS mark 5 [Aug 4 19:21:39] VERBOSE[12255] netsock.c: == Using SIP RTP CoS mark 5 I get the same error. Same random voicemail when no voicemail is configured. I was under the impressing that "s" was the catchall for all incoming trunks. What has changed? Joe -- _ -- 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] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used
On Wed, Aug 4, 2010 at 8:52 PM, Joe Wood wrote: > Hello. > > I have been beating my head over this problem for about 6 hours now. > > I have a SIP peer, who I register to (successfully), who should be > directing all incoming calls at my [default] stanza in my > extensions.conf: > > [ Context 'default' created by 'pbx_config' ] > 's' =>1. Wait(1) > [pbx_config] >2. Answer() > [pbx_config] >3. Background(welcome) > [pbx_config] >4. Background(and) > [pbx_config] >5. Background(thank-you-for-calling) > [pbx_config] >6. Background(conference-reservations) > [pbx_config] >7. Waitfor() > [pbx_config] >8. Hangup() > [pbx_config] > > Unfortunately, no matter how I configure extensions.conf or sip.conf, > the phone call always ends up saying: "Extension is unavailable. > Please leave your message after the tone". > > sip.conf: > > [general] > register => NPANXX:passw...@service_provider_ip > registertimeout=29 > registerattempts=0 > defaultexpiry=60 > > [DID_NUMBER] > type=peer > context=default > host=SERVICE_PROVIDER_IP > authuser=DID_NUMBER > fromuser=DID_NUMBER > fromdomain=SERVICE_PROVIDER_REALM > remotesecret=SERVICE_PROVIDER_PASSWD > secret=SERVICE_PROVIDER_PASSWD > dtmfmode=rfc2833 > disallow=all > allow=ulaw > qualify=yes > > I am attempting just to get the starting point where I can direct > users through my asterisk box, but it won't direct users to the 's' > extention, only to some voicemail box. I've removed the voicemail > config. > > My extensions.conf is tiny: > > [globals] > > [general] > > [default] > exten => s,1,Wait(1) > exten => s,n,Answer() > exten => s,n,Background(welcome) > exten => s,n,Background(and) > exten => s,n,Background(thank-you-for-calling) > exten => s,n,Background(conference-reservations) > exten => s,n,Waitfor() > exten => s,n,Hangup() > > > What am I doing wrong here? > > > > Thanks for any help you can give. > > > Joe > You don't have any extensions in your default context that match the extension that your sip peer is dialing in on. 's' is not a default extension for SIP...try using _X., and see what you get. Bump up the CLI (core set verbose 10) and then repost a failed called attempt. Some SIP providers also use a + symbol in front of their inbound calls, so you may need to use _+X., instead. -- Thanks, --Warren Selby http://www.selbytech.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
[asterisk-users] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used
Hello. I have been beating my head over this problem for about 6 hours now. I have a SIP peer, who I register to (successfully), who should be directing all incoming calls at my [default] stanza in my extensions.conf: [ Context 'default' created by 'pbx_config' ] 's' =>1. Wait(1)[pbx_config] 2. Answer() [pbx_config] 3. Background(welcome)[pbx_config] 4. Background(and)[pbx_config] 5. Background(thank-you-for-calling) [pbx_config] 6. Background(conference-reservations)[pbx_config] 7. Waitfor() [pbx_config] 8. Hangup() [pbx_config] Unfortunately, no matter how I configure extensions.conf or sip.conf, the phone call always ends up saying: "Extension is unavailable. Please leave your message after the tone". sip.conf: [general] register => NPANXX:passw...@service_provider_ip registertimeout=29 registerattempts=0 defaultexpiry=60 [DID_NUMBER] type=peer context=default host=SERVICE_PROVIDER_IP authuser=DID_NUMBER fromuser=DID_NUMBER fromdomain=SERVICE_PROVIDER_REALM remotesecret=SERVICE_PROVIDER_PASSWD secret=SERVICE_PROVIDER_PASSWD dtmfmode=rfc2833 disallow=all allow=ulaw qualify=yes I am attempting just to get the starting point where I can direct users through my asterisk box, but it won't direct users to the 's' extention, only to some voicemail box. I've removed the voicemail config. My extensions.conf is tiny: [globals] [general] [default] exten => s,1,Wait(1) exten => s,n,Answer() exten => s,n,Background(welcome) exten => s,n,Background(and) exten => s,n,Background(thank-you-for-calling) exten => s,n,Background(conference-reservations) exten => s,n,Waitfor() exten => s,n,Hangup() What am I doing wrong here? Thanks for any help you can give. Joe -- _ -- 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] Incoming SIP Calls
You are sending the call to from-sip-external which by default dumps
the call and gives the congestion message.
Go into your sip.conf and change from-sip-external to from-pstn or
change the context from-sip-external in extensions.conf to what you
want it to do.
My guess is you are using AAH.
On 1/17/06, Michael Sampson <[EMAIL PROTECTED]> wrote:
> I set up a deal with a voip provider to route calls to me via SIP. When
> the call hits my system I get a busy signal. I have a route set up
> through amp for the number (8002286573). Not sure what else I need to
> set up. This is what I get at the CLI.
>
> --
> asterisk*CLI>
> -- Executing AbsoluteTimeout("SIP/71.16.179.175-0856d708", "15") in
> new stack
> -- Set Absolute Timeout to 15
> -- Executing Congestion("SIP/71.16.179.175-0856d708", "") in new stack
> == Spawn extension (from-sip-external, 8002286573, 2) exited non-zero
> on 'SIP/71.16.179.175-0856d708'
> -- Executing AbsoluteTimeout("SIP/71.16.179.175-0856d708", "15") in
> new stack
> -- Set Absolute Timeout to 15
> -- Executing Congestion("SIP/71.16.179.175-0856d708", "") in new stack
> == Spawn extension (from-sip-external, h, 2) exited non-zero on
> 'SIP/71.16.179.175-0856d708'
> -- Executing AbsoluteTimeout("SIP/71.16.179.175-0856d708", "15") in
> new stack
> -- Set Absolute Timeout to 15
> -- Executing Congestion("SIP/71.16.179.175-0856d708", "") in new stack
> == Spawn extension (from-sip-external, 8002286573, 2) exited non-zero
> on 'SIP/71.16.179.175-0856d708'
> -- Executing AbsoluteTimeout("SIP/71.16.179.175-0856d708", "15") in
> new stack
> -- Set Absolute Timeout to 15
> -- Executing Congestion("SIP/71.16.179.175-0856d708", "") in new stack
> == Spawn extension (from-sip-external, h, 2) exited non-zero on
> 'SIP/71.16.179.175-0856d708'
> -- Executing AbsoluteTimeout("SIP/71.16.179.175-0856d708", "15") in
> new stack
> -- Set Absolute Timeout to 15
> -- Executing Congestion("SIP/71.16.179.175-0856d708", "") in new stack
> == Spawn extension (from-sip-external, 8002286573, 2) exited non-zero
> on 'SIP/71.16.179.175-0856d708'
> -- Executing AbsoluteTimeout("SIP/71.16.179.175-0856d708", "15") in
> new stack
> -- Set Absolute Timeout to 15
> -- Executing Congestion("SIP/71.16.179.175-0856d708", "") in new stack
> == Spawn extension (from-sip-external, h, 2) exited non-zero on
> 'SIP/71.16.179.175-0856d708'
>
> ---
>
> this is what I get in /var/log/asterisk/full
> --
> Jan 17 14:01:24 DEBUG[3033] chan_sip.c: Checking SIP call limits for device
> Jan 17 14:01:24 DEBUG[3033] chan_sip.c: build_route: Contact hop:
>
> Jan 17 14:01:24 VERBOSE[9282] logger.c: -- Executing
> AbsoluteTimeout("SIP/71.16.179.175-0856ac50", "15") in new stack
> Jan 17 14:01:24 VERBOSE[9282] logger.c: -- Set Absolute Timeout to 15
> Jan 17 14:01:24 VERBOSE[9282] logger.c: -- Executing
> Congestion("SIP/71.16.179.175-0856ac50", "") in new stack
> Jan 17 14:01:24 VERBOSE[9282] logger.c: == Spawn extension
> (from-sip-external, 8002286573, 2) exited non-zero on
> 'SIP/71.16.179.175-0856ac50'
> Jan 17 14:01:24 VERBOSE[9282] logger.c: -- Executing
> AbsoluteTimeout("SIP/71.16.179.175-0856ac50", "15") in new stack
> Jan 17 14:01:24 VERBOSE[9282] logger.c: -- Set Absolute Timeout to 15
> Jan 17 14:01:24 VERBOSE[9282] logger.c: -- Executing
> Congestion("SIP/71.16.179.175-0856ac50", "") in new stack
> Jan 17 14:01:24 VERBOSE[9282] logger.c: == Spawn extension
> (from-sip-external, h, 2) exited non-zero on 'SIP/71.16.179.175-0856ac50'
> Jan 17 14:01:24 DEBUG[9282] cdr_addon_mysql.c: cdr_mysql: inserting a
> CDR record.
> Jan 17 14:01:24 DEBUG[9282] cdr_addon_mysql.c: cdr_mysql: SQL command as
> follows: INSERT INTO cdr
> (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid)
> VALUES ('2006-01-17
> 14:01:24','6124322250','6124322250','8002286573','from-sip-external',
> 'SIP/71.16.179.175-0856ac50','','Congestion','',0,0,'NO
> ANSWER',3,'','1137528084.126')
> Jan 17 14:01:24 DEBUG[9282] chan_sip.c: update_call_counter() -
> decrement call limit counter
> Jan 17 14:01:24 DEBUG[3033] chan_sip.c: Stopping retransmission on
> '[EMAIL PROTECTED]' of Response 101:
> Match Not Found
> Jan 17 14:01:24 DEBUG[3033] chan_sip.c: Checking SIP call limits for device
> Jan 17 14:01:24 DEBUG[3033] chan_sip.c: build_route: Contact hop:
>
> Jan 17 14:01:24 VERBOSE[9286] logger.c: -- Executing
> AbsoluteTimeout("SIP/71.16.179.175-0856ac50", "15") in new stack
> Jan 17 14:01:24 VERBOSE[9286] logger.c: -- Set Absolute Timeout to 15
> Jan 17 14:01:24 VERBOSE[9286] logger.c: -- Executing
> Congestion("SIP/71.16.179.175-0856ac50", "") in new stack
> Jan 17 14:01:24 VERBOSE[9286] logger.c: == Spawn extension
> (from-sip-external, 8002286573, 2) exited non-zero on
> 'SIP/7
[Asterisk-Users] Incoming SIP Calls
I set up a deal with a voip provider to route calls to me via SIP. When
the call hits my system I get a busy signal. I have a route set up
through amp for the number (8002286573). Not sure what else I need to
set up. This is what I get at the CLI.
--
asterisk*CLI>
-- Executing AbsoluteTimeout("SIP/71.16.179.175-0856d708", "15") in
new stack
-- Set Absolute Timeout to 15
-- Executing Congestion("SIP/71.16.179.175-0856d708", "") in new stack
== Spawn extension (from-sip-external, 8002286573, 2) exited non-zero
on 'SIP/71.16.179.175-0856d708'
-- Executing AbsoluteTimeout("SIP/71.16.179.175-0856d708", "15") in
new stack
-- Set Absolute Timeout to 15
-- Executing Congestion("SIP/71.16.179.175-0856d708", "") in new stack
== Spawn extension (from-sip-external, h, 2) exited non-zero on
'SIP/71.16.179.175-0856d708'
-- Executing AbsoluteTimeout("SIP/71.16.179.175-0856d708", "15") in
new stack
-- Set Absolute Timeout to 15
-- Executing Congestion("SIP/71.16.179.175-0856d708", "") in new stack
== Spawn extension (from-sip-external, 8002286573, 2) exited non-zero
on 'SIP/71.16.179.175-0856d708'
-- Executing AbsoluteTimeout("SIP/71.16.179.175-0856d708", "15") in
new stack
-- Set Absolute Timeout to 15
-- Executing Congestion("SIP/71.16.179.175-0856d708", "") in new stack
== Spawn extension (from-sip-external, h, 2) exited non-zero on
'SIP/71.16.179.175-0856d708'
-- Executing AbsoluteTimeout("SIP/71.16.179.175-0856d708", "15") in
new stack
-- Set Absolute Timeout to 15
-- Executing Congestion("SIP/71.16.179.175-0856d708", "") in new stack
== Spawn extension (from-sip-external, 8002286573, 2) exited non-zero
on 'SIP/71.16.179.175-0856d708'
-- Executing AbsoluteTimeout("SIP/71.16.179.175-0856d708", "15") in
new stack
-- Set Absolute Timeout to 15
-- Executing Congestion("SIP/71.16.179.175-0856d708", "") in new stack
== Spawn extension (from-sip-external, h, 2) exited non-zero on
'SIP/71.16.179.175-0856d708'
---
this is what I get in /var/log/asterisk/full
--
Jan 17 14:01:24 DEBUG[3033] chan_sip.c: Checking SIP call limits for device
Jan 17 14:01:24 DEBUG[3033] chan_sip.c: build_route: Contact hop:
Jan 17 14:01:24 VERBOSE[9282] logger.c: -- Executing
AbsoluteTimeout("SIP/71.16.179.175-0856ac50", "15") in new stack
Jan 17 14:01:24 VERBOSE[9282] logger.c: -- Set Absolute Timeout to 15
Jan 17 14:01:24 VERBOSE[9282] logger.c: -- Executing
Congestion("SIP/71.16.179.175-0856ac50", "") in new stack
Jan 17 14:01:24 VERBOSE[9282] logger.c: == Spawn extension
(from-sip-external, 8002286573, 2) exited non-zero on
'SIP/71.16.179.175-0856ac50'
Jan 17 14:01:24 VERBOSE[9282] logger.c: -- Executing
AbsoluteTimeout("SIP/71.16.179.175-0856ac50", "15") in new stack
Jan 17 14:01:24 VERBOSE[9282] logger.c: -- Set Absolute Timeout to 15
Jan 17 14:01:24 VERBOSE[9282] logger.c: -- Executing
Congestion("SIP/71.16.179.175-0856ac50", "") in new stack
Jan 17 14:01:24 VERBOSE[9282] logger.c: == Spawn extension
(from-sip-external, h, 2) exited non-zero on 'SIP/71.16.179.175-0856ac50'
Jan 17 14:01:24 DEBUG[9282] cdr_addon_mysql.c: cdr_mysql: inserting a
CDR record.
Jan 17 14:01:24 DEBUG[9282] cdr_addon_mysql.c: cdr_mysql: SQL command as
follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid)
VALUES ('2006-01-17
14:01:24','6124322250','6124322250','8002286573','from-sip-external',
'SIP/71.16.179.175-0856ac50','','Congestion','',0,0,'NO
ANSWER',3,'','1137528084.126')
Jan 17 14:01:24 DEBUG[9282] chan_sip.c: update_call_counter() -
decrement call limit counter
Jan 17 14:01:24 DEBUG[3033] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' of Response 101:
Match Not Found
Jan 17 14:01:24 DEBUG[3033] chan_sip.c: Checking SIP call limits for device
Jan 17 14:01:24 DEBUG[3033] chan_sip.c: build_route: Contact hop:
Jan 17 14:01:24 VERBOSE[9286] logger.c: -- Executing
AbsoluteTimeout("SIP/71.16.179.175-0856ac50", "15") in new stack
Jan 17 14:01:24 VERBOSE[9286] logger.c: -- Set Absolute Timeout to 15
Jan 17 14:01:24 VERBOSE[9286] logger.c: -- Executing
Congestion("SIP/71.16.179.175-0856ac50", "") in new stack
Jan 17 14:01:24 VERBOSE[9286] logger.c: == Spawn extension
(from-sip-external, 8002286573, 2) exited non-zero on
'SIP/71.16.179.175-0856ac50'
Jan 17 14:01:24 VERBOSE[9286] logger.c: -- Executing
AbsoluteTimeout("SIP/71.16.179.175-0856ac50", "15") in new stack
Jan 17 14:01:24 VERBOSE[9286] logger.c: -- Set Absolute Timeout to 15
Jan 17 14:01:24 VERBOSE[9286] logger.c: -- Executing
Congestion("SIP/71.16.179.175-0856ac50", "") in new stack
Jan 17 14:01:24 VERBOSE[9286] logger.c: == Spawn extension
(from-sip-external, h, 2) exited non-zero on 'SIP/71.16.179.175-0856ac50'
Jan 17 14:01:24 DEBUG[9286] cdr
Re: [Asterisk-Users] Incoming SIP calls with no extension
Hello Christoph,
On Sat, 4 Jun 2005, Christoph Weber wrote:
> Hi All!
>
> I am new to asterisk and have a simple question:
>
> I was able to install and configure it as I wanted. But when I try to
> configure a default extension ('s' extension) for incoming sip calls it
> doesn't work. I just want that when someone calls my ip it get's
> connected to some default extension.
>
> The reason I ask is that I plan to replace a conventional pbx with
> asterisk. The setup should include a BRI interface from the telekom
> provider. I need to use the same numbers as now because they are well
> known. At the moment I just installed asterisk on my laptop to "play"
> with the configuration, and I am afraid that a caller has to use some
> extension to call in from the BRI interface when I replace the pbx, just
> like now with the software sip phone.
Set
immediate=no
overlapdial=yes
in zapata.conf
Regards
Torsten
>
> Thanks,
> Christoph
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>
--
Media Online Internet Services & Marketing GmbH
Torsten Krueger [EMAIL PROTECTED]
fon: 49-231-5575100fax: 49-231-55751098
Kurze Str. 10 D-44137 Dortmund
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[Asterisk-Users] Incoming SIP calls with no extension
Hi All!
I am new to asterisk and have a simple question:
I was able to install and configure it as I wanted. But when I try to
configure a default extension ('s' extension) for incoming sip calls it
doesn't work. I just want that when someone calls my ip it get's
connected to some default extension.
The reason I ask is that I plan to replace a conventional pbx with
asterisk. The setup should include a BRI interface from the telekom
provider. I need to use the same numbers as now because they are well
known. At the moment I just installed asterisk on my laptop to "play"
with the configuration, and I am afraid that a caller has to use some
extension to call in from the BRI interface when I replace the pbx, just
like now with the software sip phone.
Thanks,
Christoph
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[Asterisk-Users] Incoming SIP calls with different signaling and RTP IP addresses
Hello, I use Asterisk CVS-v1-0-12/21/04-11:05:29 and I noticed that when we receive calls from a partner's IP address (who has a static host entry in the sip.conf file) but the RTP comes from a different address than the signaling, our * sends a 403 forbidden message and drops the call. This problem does not llow us to receive calls from SIP proxies. Was this fixed in newer versions of Asterisk? Best regards, Vlasis. ___ 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] Incoming SIP calls not being sent to "s" extension
I was troubleshooting a problem with incoming calls to my VoicePulse Open Access (NOT Connect) numbers not coming in and I noticed the following in the SIP debug... >Found peer 'roamer1-vpoa' >Looking for s00** in ivr-incoming Why are the calls getting sent to this weird "s00**" extension and not the usual "s" extension in context ivr-incoming as they should? Of course, that "s00" extension happens to be the username for my VP Open Access account; I'm thinking that the first letter being an "s" is confusing Asterisk, since I'm not seeing the same thing with FWD or other services where the username is all numbers. relevant parts of sip.conf for the VP OA account: register => s00**:[EMAIL PROTECTED] ; [roamer1-vpoa] type=friend context=ivr-incoming username=s00** secret=SeCrEt host=access1.voicepulse.com dtmf=inband nat=yes qualify=yes canreinvite=no insecure=very My FWD, SIPPhone, etc. accounts are configured *exactly* the same except for different usernames, secrets, and hosts and they work fine...calls to those numbers go to the "s" extension in context ivr-incoming as they should. I did come up with a workaround (make a special context contaning the "s00" username as an extension that just Gotos the s extension in ivr-incoming), but I shouldn't have to do that... -SC -- Stanley Cline -- sc1 at roamer1 dot org -- http://www.roamer1.org/ ... "Never put off until tomorrow what you can do today. There might be a law against it by that time." -/usr/games/fortune ___ 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] Incoming SIP calls as asterisk@...
Hi all, I noticed that all incoming calls come from the user "[EMAIL PROTECTED]", so I just can't hit the "Call" button on my SJphone for Linux to return the call... Is there any way to configure Asterisk to show the real [EMAIL PROTECTED] ? Thanks and regards, Martin ___ 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] incoming SIP calls drop on pickup.
I also thought it might be a coded mismatch. Maybe someone can explain why outgoing calls work when incoming calls between the same phones don't work? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, March 29, 2004 10:32 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] incoming SIP calls drop on pickup. Sounds like a codec mismatch to me. I had a similar problem with ICH. On Mon, 29 Mar 2004 19:23:15 +0100, "jc" wrote: Hi All, I have an annoying problem. Out going SIP/sipphone.com calls work fine. Internal calls work fine. However, incoming SIP calls DIAL and ring, but send a busy signal when picked up. The same happens if I take the SNOM200 out of the loop and just try to answer and playback a recording. The debugs don't produce an obvious error. Thanks JC ___ 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] incoming SIP calls drop on pickup.
Sounds like a codec mismatch to me. I had a similar problem with ICH. On Mon, 29 Mar 2004 19:23:15 +0100, "jc" wrote: Hi All, I have an annoying problem. Out going SIP/sipphone.com calls work fine. Internal calls work fine. However, incoming SIP calls DIAL and ring, but send a busy signal when picked up. The same happens if I take the SNOM200 out of the loop and just try to answer and playback a recording. The debugs dont produce an obvious error. Thanks JC ___ 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] incoming SIP calls drop on pickup.
Hi All, I have an annoying problem. Out going SIP/sipphone.com calls work fine. Internal calls work fine. However, incoming SIP calls DIAL and ring, but send a busy signal when picked up. The same happens if I take the SNOM200 out of the loop and just try to answer and playback a recording. The debugs don’t produce an obvious error. Thanks JC
[Asterisk-Users] Incoming SIP calls
Hello All I am trying to answer incoming SIP calls, first, by dialing an extension, thence into voicemail, which works; and secondly by going straight into voice mail which does not. The extension.conf that works is like this; [incomingSIP] exten=>_.,1,Dial,Zap/2|1 exten=>_.,2,Voicemail,u5152 exten=>_.,3,Hangup the extension.conf which does not is like this; [incomingSIP] exten=>_.,1,Answer exten=>_.,2,Voicemail,u5152 exten=>_.,3,Hangup For the non-working config I cam see the commands being run on the console but the SIP session times out without receiving any audio. I have traced both sessions with ethereal and the protocol handshake is identical however * appears to be ignoring the ACK response for the second config and repeatedly sends 200/OK and then times out. Isuppose I am missing something obvious here but am going 'glassy eyed' trying to spot it. Any help appreciated. Brian ___ 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] Incoming SIP-calls and Festival
Hi!
I have problems with calls that are coming from a SIP-provider, and where I
want to use Festival to play som text to the caller.
I hear the text if I call from a SIP-extension (I've tried with g.711a/u and
GSM and all three works)
But if I call in to the server through my SIP-provider I wont hear any
Festival-speech (no error output on the console - see in the end of the
mail), if I instead use Background for example I can hear the soundfile.
I think it's very strange - is there anyone that have an idea why I can't
use Festival with the calls coming from my SIP-provider.
This is how it looks on the console - but the caller don't hear anything;
-- Executing Answer("SIP/11292-594f", "") in new stack
-- Executing Festival("SIP/11292-594f", "'Hello'") in new stack
== Parsing '/etc/asterisk/festival.conf': Found
== Spawn extension (digisip, 301, 2) exited non-zero on 'SIP/11292-594f'
Regards, Lars
---
Lars Fredriksson
Ockelbo, Sweden
mailto:[EMAIL PROTECTED]
http://www.fredriksson.net/
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