# Outgoing context for foo provider **URL:** \ **Category:** Asterisk Support\ **Created:** [July 2, 2016, 12:07pm UTC](https://community.asterisk.org/t/outgoing-context-for-foo-provider/67264 "2016-07-02T12:07:36Z")\ **Posts on this page:** 5\ **Page:** 1
**Author:** ![thufir](https://avatars.discourse-cdn.com/v4/letter/t/ee7513/32.png) [@thufir](https://community.asterisk.org/u/thufir)\ **Post date:** [July 2, 2016, 12:07pm UTC](https://community.asterisk.org/t/outgoing-context-for-foo-provider/67264/1 "2016-07-02T12:07:36Z")
I’m trying to understand why I can’t dial out through a SIP peer acting as a trunk. In what way are these two attempts to dial an outbound number different? The first at least reaches the peer for trunking? What’s happening, in particular, with the first call? My inclination is for three contexts: internal, inbound and outbound. I don’t quite understand why [https://www.beardy.se/how-to-set-up-a-sip-trunk-in-the-asterisk-pbx](https://www.beardy.se/how-to-set-up-a-sip-trunk-in-the-asterisk-pbx) only has `incoming` and `myphones`, with `myphones` also directing to…`foo provider`? In sip.conf `foo provider` has a context of `incoming` but then is being used to dial out…which makes my head hurt. Having difficulty following the chain of events from the last few lines of `myphones` context where `foo provider` is invoked. Good, yes, use `foo provider` to dial out…but shouldn’t `foo provider` not have an `incoming` context if it’s being used for outgoing calls? ``` mordor*CLI> mordor*CLI> channel originate SIP/thufir extension 918002506510@myphones == Using SIP RTP CoS mark 5 -- Called thufir -- SIP/thufir-0000002c is ringing -- SIP/thufir-0000002c answered -- Executing [918002506510@myphones:1] Log("SIP/thufir-0000002c", "NOTICE, Dialing out from "" <> to 18002506510 through 12345678GW1") in new stack [Jul 2 04:52:14] NOTICE[6787][C-00000026]: Ext. 918002506510:1 @ myphones: Dialing out from "" <> to 18002506510 through 12345678GW1 -- Executing [918002506510@myphones:2] Dial("SIP/thufir-0000002c", "SIP/12345678GW1/18002506510,60") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/12345678GW1/18002506510 -- SIP/12345678GW1-0000002d answered SIP/thufir-0000002c -- Channel SIP/thufir-0000002c joined 'simple_bridge' basic-bridge <6ed476b6-9c9e-4833-bc04-70e405f43dd2> -- Channel SIP/12345678GW1-0000002d joined 'simple_bridge' basic-bridge <6ed476b6-9c9e-4833-bc04-70e405f43dd2> -- Channel SIP/thufir-0000002c left 'native_rtp' basic-bridge <6ed476b6-9c9e-4833-bc04-70e405f43dd2> == Spawn extension (myphones, 918002506510, 2) exited non-zero on 'SIP/thufir-0000002c' -- Channel SIP/12345678GW1-0000002d left 'native_rtp' basic-bridge <6ed476b6-9c9e-4833-bc04-70e405f43dd2> mordor*CLI> mordor*CLI> mordor*CLI> channel originate SIP/thufir extension 918002506510@incoming == Using SIP RTP CoS mark 5 -- Called thufir -- SIP/thufir-0000002e is ringing -- SIP/thufir-0000002e answered [Jul 2 04:52:44] WARNING[6792][C-00000027]: pbx.c:6492 __ast_pbx_run: Channel 'SIP/thufir-0000002e' sent to invalid extension but no invalid handler: context,exten,priority=incoming,918002506510,1 mordor*CLI> mordor*CLI> ``` peers and trunk: ``` mordor*CLI> mordor*CLI> sip show peers Name/username Host Dyn Forcerport Comedia ACL Port Status Description 12345678GW1/42654565 63.247.69.226 Yes Yes 5060 OK (85 ms) 12345678GW2/42654565 63.247.69.226 Yes Yes 5060 OK (82 ms) demo_alice (Unspecified) D Yes Yes 0 UNKNOWN demo_bob (Unspecified) D Yes Yes 0 UNKNOWN hawat/hawat 192.168.1.6 D Yes Yes 5060 OK (410 ms) thufir/thufir 192.168.1.5 D Yes Yes 5062 OK (9 ms) 6 sip peers [Monitored: 4 online, 2 offline Unmonitored: 0 online, 0 offline] mordor*CLI> mordor*CLI> sip show peer 12345678GW1 * Name : 12345678GW1 Description : Secret : MD5Secret : Remote Secret: Context : incoming Record On feature : automon Record Off feature : automon Subscr.Cont. : Language : Tonezone : AMA flags : Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened FromDomain : gw1.siptrunk.com Port 5060 Callgroup : Pickupgroup : Named Callgr : Nam. Pickupgr: MOH Suggest : Mailbox : VM Extension : asterisk LastMsgsSent : 0/0 Call limit : 0 Max forwards : 0 Dynamic : No Callerid : "" <> MaxCallBR : 384 kbps Expire : -1 Insecure : invite Force rport : Yes Symmetric RTP: Yes ACL : No DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: 4294967295 DirectMedia : No PromiscRedir : No User=Phone : No Video Support: No Text Support : No Ign SDP ver : No Trust RPID : Yes Send RPID : No Path support : No Path : N/A TrustIDOutbnd: Legacy Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : gw1.siptrunk.com Addr->IP : 63.247.69.226:5060 Defaddr->IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: 42654565 SIP Options : (none) Codecs : (ulaw|alaw|gsm) Auto-Framing : No Status : OK (85 ms) Useragent : Reg. Contact : Qualify Freq : 30000 ms Keepalive : 0 ms Sess-Timers : Refuse Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No Encryption : No ``` dialplan: ``` mordor*CLI> mordor*CLI> dialplan show __func_periodic_hook_context__ ael-builtin-h-bubble ael-default ael-demo ael-dundi-e164 ael-dundi-e164-canonical ael-dundi-e164-customers ael-dundi-e164-local ael-dundi-e164-lookup ael-dundi-e164-switch ael-dundi-e164-via-pstn ael-iaxprovider ael-iaxtel700 ael-international ael-local ael-longdistance ael-std-exten-ael ael-trunkint ael-trunkld ael-trunklocal ael-trunktollfree chanvar default demo globals local myphones parkedcalls public incoming mordor*CLI> dialplan show myphones [Context 'myphones' created by 'pbx_config'] '1000' => 1. Dial(SIP/1000) [pbx_config] 2. Hangup() [pbx_config] '1001' => 1. Dial(SIP/1001) [pbx_config] 2. Hangup() [pbx_config] '201' => 1. Answer() [pbx_config] 2. Playback(tt-monty-knights) [pbx_config] 3. Hangup() [pbx_config] '202' => 1. Answer() [pbx_config] 2. Playback(welcome) [pbx_config] 3. Playback(demo-echotest) [pbx_config] 4. Echo() [pbx_config] 5. Playback(demo-echodone) [pbx_config] 6. Playback(vm-goodbye) [pbx_config] 7. Hangup() [pbx_config] '4000' => 1. Playback(tt-monkeys) [pbx_config] '5000' => 1. Playback(tt-monkeysintro) [pbx_config] '555' => 1. Playback(hello-world) [pbx_config] 2. Playback(echo-test) [pbx_config] 3. Echo() [pbx_config] 4. Playback(demo-echodone) [pbx_config] '6001' => 1. Dial(SIP/demo_alice,20) [pbx_config] '6002' => 1. Dial(SIP/demo_bob,20) [pbx_config] '6003' => 1. Dial(SIP/thufir,20) [pbx_config] '6004' => 1. Dial(SIP/hawat,20) [pbx_config] '_9XXXX.' => 1. Log(NOTICE, Dialing out from ${CALLERID(all)} to ${EXTEN:1} through 12345678GW1) [pbx_config] 2. Dial(SIP/12345678GW1/${EXTEN:1},60) [pbx_config] 3. Playtones(congestion) [pbx_config] 4. Hangup() [pbx_config] -= 12 extensions (28 priorities) in 1 context. =- mordor*CLI> mordor*CLI> mordor*CLI> dialplan show incoming [Context 'incoming' created by 'pbx_config'] 's' => 1. Log(NOTICE, Incoming call from ${CALLERID(all)}) [pbx_config] 2. Dial(SIP/1000) [pbx_config] 3. Hangup() [pbx_config] -= 1 extension (3 priorities) in 1 context. =- mordor*CLI> ``` I’ve tried many variations on dialing the number to match the pattern. this is largely based on: > **[How to set up a SIP trunk in the Asterisk PBX](https://www.beardy.se/how-to-set-up-a-sip-trunk-in-the-asterisk-pbx)** > > This time I will show you how to configure a SIP trunk in Asterisk, and add extensions in the dialplan so that the telephones can dial out through the trunk. ---
**Author:** ![astbox](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/astbox/32/1691_2.png) [@astbox](https://community.asterisk.org/u/astbox)\ **Post date:** [July 2, 2016, 1:13pm UTC](https://community.asterisk.org/t/outgoing-context-for-foo-provider/67264/2 "2016-07-02T13:13:22Z")
What do you want to do, why do you try to dial out from the cli? Give more details or else noone can help you. ---
**Author:** ![thufir](https://avatars.discourse-cdn.com/v4/letter/t/ee7513/32.png) [@thufir](https://community.asterisk.org/u/thufir)\ **Post date:** [July 2, 2016, 10:31pm UTC](https://community.asterisk.org/t/outgoing-context-for-foo-provider/67264/3 "2016-07-02T22:31:44Z")
I’m just trying to dial from either `thufir` or `hawat`, both attached devices. The `thufir` device is on a hardphone, the `hawat` device is on a softphone (Csip) on Android. The doc I’m also referencing is: > **[SIPTRUNK.com CONFIGURATION GUIDE FOR ASTERISK](https://support.siptrunk.com/hc/en-us/articles/204848079-SIPTRUNK-com-CONFIGURATION-GUIDE-FOR-ASTERISK)** > > We recommend you create two trunk configurations for each SIPTRUNK.COM trunk to register to each of our servers at gw1.siptrunk.com and gw2.siptrunk.com. (gw1.siptrunk.com is primary and gw2.siptru... which seems to omit the dialplan specifics. However: > **[Outbound Calling HOW TO](https://support.siptrunk.com/hc/en-us/articles/207691713-Outbound-Calling-HOW-TO)** > > Do you find yourself asking the question “why doesn't my call work?” You may even be able to see your call in the SIPTRUNK.com CDR page! Still you struggle with comprehending why your outbound call... says that they just need ten digits, basically. I was on the CLI trying to initiate calls with variations to find out which worked. Prepend with a 9, don’t prepend, etc. My conclusion was that there’s something wrong, most likely, with the dialplan, specifically relating to the `myphones` context. ---
**Author:** ![david551](https://avatars.discourse-cdn.com/v4/letter/d/3bc359/32.png) [@david551](https://community.asterisk.org/u/david551)\ **Post date:** [July 2, 2016, 10:43pm UTC](https://community.asterisk.org/t/outgoing-context-for-foo-provider/67264/4 "2016-07-02T22:43:00Z")
Rather than trying to put a system together from half understood examples, please go to [http://www.asteriskdocs.org/](http://www.asteriskdocs.org/) and read up how Asterisk configurations really work. ---
**Author:** ![thufir](https://avatars.discourse-cdn.com/v4/letter/t/ee7513/32.png) [@thufir](https://community.asterisk.org/u/thufir)\ **Post date:** [July 3, 2016, 6:59am UTC](https://community.asterisk.org/t/outgoing-context-for-foo-provider/67264/5 "2016-07-03T06:59:27Z")
Seem to have solved it to a degree in by using `include => outgoing` in `myphones` so that there’s not any duplicated code. ``` mordor*CLI> mordor*CLI> channel originate SIP/thufir extension 18003569377@outgoing == Using SIP RTP CoS mark 5 -- Called thufir -- SIP/thufir-0000002e is ringing -- SIP/thufir-0000002e answered -- Executing [18003569377@outgoing:1] NoOp("SIP/thufir-0000002e", "") in new stack -- Executing [18003569377@outgoing:2] Log("SIP/thufir-0000002e", "NOTICE, Dialing out from "" <> to 8003569377 through TELNYX") in new stack [Jul 3 00:23:36] NOTICE[5328][C-00000022]: Ext. 18003569377:2 @ outgoing: Dialing out from "" <> to 8003569377 through TELNYX -- Executing [18003569377@outgoing:3] Dial("SIP/thufir-0000002e", "SIP/TELNYX/8003569377,60") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/TELNYX/8003569377 > 0x7f25a0060900 -- Probation passed - setting RTP source address to 192.168.1.5:16390 -- SIP/TELNYX-0000002f is ringing -- SIP/TELNYX-0000002f is making progress passing it to SIP/thufir-0000002e > 0x7f2590008a80 -- Probation passed - setting RTP source address to 64.16.240.36:27938 -- SIP/TELNYX-0000002f answered SIP/thufir-0000002e -- Channel SIP/thufir-0000002e joined 'simple_bridge' basic-bridge <8dc1f203-8fc8-4791-bc5b-0f1517b9449d> -- Channel SIP/TELNYX-0000002f joined 'simple_bridge' basic-bridge <8dc1f203-8fc8-4791-bc5b-0f1517b9449d> > Bridge 8dc1f203-8fc8-4791-bc5b-0f1517b9449d: switching from simple_bridge technology to native_rtp -- Channel SIP/thufir-0000002e left 'native_rtp' basic-bridge <8dc1f203-8fc8-4791-bc5b-0f1517b9449d> -- Channel SIP/TELNYX-0000002f left 'native_rtp' basic-bridge <8dc1f203-8fc8-4791-bc5b-0f1517b9449d> == Spawn extension (outgoing, 18003569377, 3) exited non-zero on 'SIP/thufir-0000002e' mordor*CLI> == Using SIP RTP CoS mark 5 -- Executing [18003563977@myphones:1] NoOp("SIP/thufir-00000030", "") in new stack -- Executing [18003563977@myphones:2] Log("SIP/thufir-00000030", "NOTICE, Dialing out from "thufir" to 8003563977 through TELNYX") in new stack [Jul 3 00:23:55] NOTICE[5330][C-00000023]: Ext. 18003563977:2 @ myphones: Dialing out from "thufir" to 8003563977 through TELNYX -- Executing [18003563977@myphones:3] Dial("SIP/thufir-00000030", "SIP/TELNYX/8003563977,60") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/TELNYX/8003563977 -- SIP/TELNYX-00000031 is ringing -- SIP/TELNYX-00000031 is making progress passing it to SIP/thufir-00000030 -- SIP/TELNYX-00000031 answered SIP/thufir-00000030 -- Channel SIP/thufir-00000030 joined 'simple_bridge' basic-bridge <9a346de5-64eb-480b-8d6c-d0253fceeb7f> -- Channel SIP/TELNYX-00000031 joined 'simple_bridge' basic-bridge <9a346de5-64eb-480b-8d6c-d0253fceeb7f> > Bridge 9a346de5-64eb-480b-8d6c-d0253fceeb7f: switching from simple_bridge technology to native_rtp > 0x7f25d0009dd0 -- Probation passed - setting RTP source address to 192.168.1.5:16392 > 0x7f259000ae20 -- Probation passed - setting RTP source address to 64.16.240.36:19754 -- Channel SIP/thufir-00000030 left 'native_rtp' basic-bridge <9a346de5-64eb-480b-8d6c-d0253fceeb7f> -- Channel SIP/TELNYX-00000031 left 'native_rtp' basic-bridge <9a346de5-64eb-480b-8d6c-d0253fceeb7f> == Spawn extension (myphones, 18003563977, 3) exited non-zero on 'SIP/thufir-00000030' mordor*CLI> mordor*CLI> mordor*CLI> dialplan show __func_periodic_hook_context__ ael-builtin-h-bubble ael-default ael-demo ael-dundi-e164 ael-dundi-e164-canonical ael-dundi-e164-customers ael-dundi-e164-local ael-dundi-e164-lookup ael-dundi-e164-switch ael-dundi-e164-via-pstn ael-iaxprovider ael-iaxtel700 ael-international ael-local ael-longdistance ael-std-exten-ael ael-trunkint ael-trunkld ael-trunklocal ael-trunktollfree chanvar default demo globals local outgoing parkedcalls public myphones mordor*CLI> dialplan show myphones [Context 'myphones' created by 'pbx_config'] '1000' => 1. Dial(SIP/1000) [pbx_config] 2. Hangup() [pbx_config] '1001' => 1. Dial(SIP/1001) [pbx_config] 2. Hangup() [pbx_config] '201' => 1. Answer() [pbx_config] 2. Playback(tt-monty-knights) [pbx_config] 3. Hangup() [pbx_config] '202' => 1. Answer() [pbx_config] 2. Playback(welcome) [pbx_config] 3. Playback(demo-echotest) [pbx_config] 4. Echo() [pbx_config] 5. Playback(demo-echodone) [pbx_config] 6. Playback(vm-goodbye) [pbx_config] 7. Hangup() [pbx_config] '4000' => 1. Playback(tt-monkeys) [pbx_config] '5000' => 1. Playback(tt-monkeysintro) [pbx_config] '555' => 1. Playback(hello-world) [pbx_config] 2. Playback(echo-test) [pbx_config] 3. Echo() [pbx_config] 4. Playback(demo-echodone) [pbx_config] '6001' => 1. Dial(SIP/demo_alice,20) [pbx_config] '6002' => 1. Dial(SIP/demo_bob,20) [pbx_config] '6003' => 1. Dial(SIP/thufir,20) [pbx_config] '6004' => 1. Dial(SIP/hawat,20) [pbx_config] Include => 'outgoing' [pbx_config] -= 11 extensions (24 priorities) in 1 context. =- mordor*CLI> mordor*CLI> dialplan show outgoing [Context 'outgoing' created by 'pbx_config'] '_1NXXNXXXXXX' => 1. NoOp() [pbx_config] 2. Log(NOTICE, Dialing out from ${CALLERID(all)} to ${EXTEN:1} through TELNYX) [pbx_config] 3. Dial(SIP/TELNYX/${EXTEN:1},60) [pbx_config] 4. Playtones(congestion) [pbx_config] 5. Hangup() [pbx_config] -= 1 extension (5 priorities) in 1 context. =- mordor*CLI> ``` Not sure that this is at all robust, but works for at least one test call, and allows dialing from the hard phone. Still don’t know how to configure the `incoming` context, however.