| 00:00.29 | francois_ | lanning: mhhh, i mean that using the default config the dtmf doesn't arrive at the other except when using the ulaw codec |
| 00:01.02 | lanning | what is generating the dtmf? |
| 00:01.03 | francois_ | other end |
| 00:02.16 | francois_ | my setup is like that: cisco SIP phone <-- SIP ulaw ---> Asterisk <--- IAX gsm ----> Asterisk <---- IAX ulaw ----> PSTN |
| 00:02.47 | lanning | The cisco phone is generating the dtmf? |
| 00:02.57 | francois_ | yes |
| 00:03.02 | file | set everything for rfc2833, problem solved |
| 00:03.08 | Damin | francois_: Use Urfc2833 |
| 00:03.15 | lanning | yup |
| 00:03.28 | francois_ | in iax.conf ? dtmfmode=rfc2833 ? |
| 00:03.34 | Damin | francois_: You can set that on the phone, and on both IAX and SIP files.. |
| 00:04.02 | Damin | Actually.. |
| 00:04.12 | Damin | Do you set dtmfmode in iax.conf? |
| 00:04.38 | lanning | I don't think so. |
| 00:05.05 | francois_ | actually that's my problem :) |
| 00:05.50 | *** join/#asterisk iMediax (iMediax@00d0a8003aa3.click-network.com) |
| 00:05.56 | lanning | the Cisco phone and the sip.conf in the asterisk server it talks to needs to use rfc2833 for dtmf. at that point it will stay OOB until it is not possible (ie. it hits the PSTN) |
| 00:07.10 | francois_ | thanks, i'll check |
| 00:07.54 | francois_ | btw what's the difference between dtmfmode=info and dtmfmode=rfc2833 ? |
| 00:08.19 | lanning | where in the sip packet the DTMF digit is placed. |
| 00:08.31 | file | rfc2833 is what I always recommend, and I love it |
| 00:09.05 | file | I am in no way paid by rfc2833 to say that. |
| 00:09.06 | file | Honest. |
| 00:09.14 | lanning | :) |
| 00:09.18 | francois_ | Yeah ! Thanks a lot, it works ! |
| 00:10.26 | lanning | It is getting about time for me to get a Cisco phone.... hmmm... now money, money, money... |
| 00:11.00 | francois_ | lanning: did you had a look on ebay ? |
| 00:11.34 | lanning | ya, I just have a current investment in a couple of channelbanks. |
| 00:12.04 | lanning | Along with the Cisco phones, I would need to rewire my house. |
| 00:13.10 | *** join/#asterisk r0d3nt-lv (~RatMan@wsip-24-234-241-186.lv.lv.cox.net) |
| 00:13.26 | francois_ | since i bought a grandstream handytone with a dect phone, i don't use my cisco very often |
| 00:13.45 | *** join/#asterisk |Vulture| (~Vulture@adsl-218-222-95.jax.bellsouth.net) |
| 00:13.59 | |Vulture| | Anyone here use Vonage with Asterisk? |
| 00:16.44 | file | |Vulture|: okay here we go again... you can't use Vonage directly with Asterisk |
| 00:16.50 | file | |Vulture|: Vonage trys very hard to prevent that |
| 00:16.54 | |Vulture| | thats what I thought |
| 00:17.00 | file | |Vulture|: however, you can get an FXO card or whatever and hook their ATA into it |
| 00:17.10 | lanning | A friend of mine used to have the ATA186(Vonage) connected to a couple of FXO cards. Eventualy he moved to a different provider. |
| 00:17.30 | |Vulture| | lanning: what provider did he use? I looked into VoicePulse |
| 00:17.48 | file | Voicepulse Connect can be used with asterisk, Voicepulse cann not |
| 00:17.51 | file | er can not |
| 00:17.56 | lanning | That is what he used. (He eventualy got IAX connected to them.) |
| 00:19.53 | |Vulture| | file: do you know of any providers that support it with an Unlimited local/LD plan? |