I’ve tried using the current SVN trunk of asterisk and asterisk-addons in order to get my hands on chan_mobile but sadly asterisk just doesn’t not work, in particular I can no longer call other extensions, I get:
[size=75][Aug 9 13:11:51] VERBOSE[2286] logger.c: – Executing [s@macro-dial:11] Dial(“SIP/101-0877ab58”, “SIP/102|15|Ttr”) in new stack
[Aug 9 13:11:51] DEBUG[2286] chan_sip.c: Asked to create a SIP channel with formats: 0x4 (ulaw)
[Aug 9 13:11:51] VERBOSE[2286] logger.c: == Using TOS bits 0
[Aug 9 13:11:51] VERBOSE[2286] logger.c: == Using CoS mark 5
[Aug 9 13:11:51] DEBUG[2286] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Aug 9 13:11:51] WARNING[2286] chan_sip.c: No such host: 102|15|Ttr
[Aug 9 13:11:51] DEBUG[2286] chan_sip.c: Cant create SIP call - target device not registred
[Aug 9 13:11:51] DEBUG[2286] chan_sip.c: Destroying SIP dialog 75409c564c7acd136eab35e42f9b2fbc@192.168.1.16
[Aug 9 13:11:51] WARNING[2286] app_dial.c: Unable to create channel of type ‘SIP’ (cause 3 - No route to destination)
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I tried the Developers forum here but that contains a single post with an invalid link.
Be forewarned that they’re very specific about not providing user-level support on that list. So, if you’ve run into an issue you’re confident is a bug or some other code related issue, please feel free.
Bug reports themselves should be directed to the bug/issue tracker at:
Be forewarned that they’re very specific about not providing user-level support on that list. So, if you’ve run into an issue you’re confident is a bug or some other code related issue, please feel free.
Bug reports themselves should be directed to the bug/issue tracker at:
I guess it’s a fine line between being keen to test out/use the new features and understanding completely what you are doing. As soon as you get down to code level it’s a big beast… I think I will just have to sit on my hands and wait for chan_mobile to hit a release.
This seems like changing in argument order or number of dial.
In my case, I was able to dial with asterisk svn yesterday.
I used very very simple extensions.conf (in my case iax2 voip phone and nokia mobile):
[Aug 9 13:11:51] WARNING[2286] chan_sip.c: No such host: 102|15|Ttr
[/quote]
This seems like changing in argument order or number of dial.
In my case, I was able to dial with asterisk svn yesterday.
I used very very simple extensions.conf (in my case iax2 voip phone and nokia mobile):
[/quote]
Thanks Phokz,
I think something a little more fundamental is screwed up (like my brain) - although I couldn’t see what - I couldn’t see any changes in dial.c or chan_sip.c other than some DNS management stuff and the like.
I pulled the rev just before they changed to using ast_debug and that compiled (and works!) OK with 1.4.10 so I’m happy about that and even happier that you posted your incoming context - that’s one less thing for me to think about… now where did I put that spare phone that happens to have bluetooth and supports SMS retrieval!!
habile: I am looking forward to read about succesfully finding your mobile and got it working.
Meanwhile I have still some issues with random one way totaly distorted audio. I have allready tried to upgrade bluez-libs, tried 2 different kernels,
tried several versions of asterisk 1.4.5-1.4.10. Not sure what to try next.
chan_mobile.c: In function 'mbl_load_config':
chan_mobile.c:1816: error: too many arguments to function 'ast_config_load'
make[1]: *** [chan_mobile.o] Fehler 1
[quote=“habile”]I’ve tried using the current SVN trunk of asterisk and asterisk-addons in order to get my hands on chan_mobile but sadly asterisk just doesn’t not work, in particular I can no longer call other extensions, I get:
[size=75][Aug 9 13:11:51] VERBOSE[2286] logger.c: – Executing [s@macro-dial:11] Dial(“SIP/101-0877ab58”, “SIP/102|15|Ttr”) in new stack
[Aug 9 13:11:51] DEBUG[2286] chan_sip.c: Asked to create a SIP channel with formats: 0x4 (ulaw)
[Aug 9 13:11:51] VERBOSE[2286] logger.c: == Using TOS bits 0
[Aug 9 13:11:51] VERBOSE[2286] logger.c: == Using CoS mark 5
[Aug 9 13:11:51] DEBUG[2286] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Aug 9 13:11:51] WARNING[2286] chan_sip.c: No such host: 102|15|Ttr
[Aug 9 13:11:51] DEBUG[2286] chan_sip.c: Cant create SIP call - target device not registred
[Aug 9 13:11:51] DEBUG[2286] chan_sip.c: Destroying SIP dialog 75409c564c7acd136eab35e42f9b2fbc@192.168.1.16
[Aug 9 13:11:51] WARNING[2286] app_dial.c: Unable to create channel of type ‘SIP’ (cause 3 - No route to destination)
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I tried the Developers forum here but that contains a single post with an invalid link.
Am I just being stupid?[/quote]
You’re not being stupid. The argument list format has changed. Pipes no longer seem to work. Change them to commas and it should work. Eg:
- that’s one less thing for me to think about… now where did I put that spare phone that happens to have bluetooth and supports SMS retrieval!! 