LIVE CALL( Green ) but no sound--- logs checking

[mg1]

I cant hear anything on client machine after being registered with my SIP provider

I get following messages

[26485]: Chan_sip.c:2608 sip_write:asked to transmit frame type 64 , while native formats is 256 ) (read/write=256/256)

[ :channel.c:2781 ast_channel_make_compatible: no path to translate from local/8600051@default-73cd,1(164) to SIP/siptrunk-083b90c0(256)

app_dial.c:16218 dial_exec_full: Had to drop call because I couldn't make local/8600051@default-73cd,1 compatible with SIP/siptrunk-083b90c0.


Kindly tell me how may I check call log made by vicidial. or any other log .. THE log that has been generated at asterisk - r prompt.

[brett05]

i'am sure that this probleme come from the codec .
first say us what version you have ?
vicidial version/
asterisk version ?
your dialplan ?

[mg1]

Thanks for quick response .. it is

Asterisk 1.2.30.2
vicidialnow ce 1.3
kernel 8.6.18xxxxx

kindly tell me how may I resolve this codec issue.

thanks

[brett05]

after vicidialnow is installed have you try to update ?
then what kind of codec you receive from your operator and what dialplan do you use ?

[mg1]

I did following config

register => 4037709321:adam@93.89.80.187

[siptrunk]
type=friend
username=4037709321
secret=adam
host=93.89.80.187
dtmfmode=rfc2833
context=inbound
canreinvite=no
disallow=all
allow=g729
insecure=port,invite

are you trying to say that I need to install a free codec for test purpose? or you are talking about something else.. do let me know.

http://www.eflo.net/VICIDIALforum/viewt ... 39f1aabd5a

thankyou

[brett05]

the free codec may work also but if you choose the commercial one it will be more good or compile the ippi one

[raju]

hi,

even i have the same problem. it is showing green live call. but no sound.

channel.c:2781 ast_channel_make_compatible: No path to translate from Local/8600051@default-60be,1(64) to SIP/siptrunk-0875a2e0(256)
app_dial.c:1628 dial_exec_full: Had to drop call because I couldn't make Local/8600051@default-60be,1 compatible with SIP/siptrunk-0875a2e0

this wornings comes back.

please help me

[williamconley]

Raju,

Please start your own thread instead of rejuvenating one that is over two years old.

And when you do start your own thread:

when you post, please post your entire configuration including (but not limited to) your installation method and vicidial version with build.

this IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)

You should also post: Asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box. If your installation method is "from scratch" you must post your operating system and should also post the .iso version from which you installed your original operating system. If your installation is "Hosted" list the site name of the host.

If this is a "Cloud" or "Virtual" server, please note the technology involved along with the version of that techology (ie: VMware Server Version 2.0.2). If it is not, merely stating the Motherboard model # and CPU would be helpful.

Similar to This:

Vicibox X.X from .iso | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel DG35EC | Core2Quad Q6600

Additionally: If your question is related to this thread, it would be because you have "allow=g729" but have not INSTALLED g729. In which case, do not "allow=g729" but rather "allow=ulaw". g729 is not free, ulaw is free and is already included in your distro.