Iam lookong for an Softphone for iPhor oder Android smartphone using togehter with an headset. I tried Zoiper and CSipSimple but quality was bad compared to an desktop SIP phone.Is there an better softphone?Or are there softphone solutions for PC desk..
Calling linphone from asterisk 13.9.1.:Dial(SIPemail@example.com)And it works. But on the linphone side the caller is:@firstname.lastname@example.orgIs there any way to make it more descriptive, at least for the sip user name ? I tried setting SIPCALL..
On 2012-05-25 16:47, jose carlos de souza wrote: > > > I need help. > > Sorry for my bad english, im from Brasil. And just starting to leran > Linux because of Asterisk. > > I want to configure Asterisk 1.8 with ipv6 in my Lan network at home. > > ..
Hi Im looking for a SIP client for Mac OS X (Im running Lion) that has video support. Ive tried Linphone but for the life of me I cant get it to add a sip account (the apply button is always grayed out)…. 🙁 Can anyone recommend other SIP clie..
On 08/26/2011 02:02 PM, linux guy wrote: get any cheap android device and load linphone. or grandstream works for a wired device. gxp2000 has enough line buttons you can easily route calls for multiple people to a phone and tell who the call is fo..
all, Ive got something strange, that got me searching for quite awhile. Configuration as followed: Linphone on a laptop, that is connected via openvpn to a proxy. That proxy is connected with iax to another asterisk. On the second one i have seve..
Is anybody here familiar with the meaning of INVAL packets for IAX2? Every few days I get a dropped outgoing call in the middle of the conversation (the outgoing call has been connected for few minutes) when an incoming call comes in. The log reads ..
I dial on A* from a linphonec to a Playback() extension, then suddenly the sound stops after a while, without any notice. I enabled debug both in linphone and A*, and the RTP packets are sent from A* and received from linphone. It doesnt matter whet..
Well the problem seems to be: the linphones are listening on port 5062, while * is on port 5060. For some reason, the INVITEs are received from *, but are forwarded on port 5060 by default. I solved the problem by moving * to port 5062 and moving ..
all, please help… I am calling in the simplest way among two linphone clients connected to one asterisk server… the call ends on one side without any sign of problem, while on the other side it stays connected. I checked the SIP dialogue and at s..