similar to: help with annoying warning message: Asked to transmit frame type ulaw, while native formats is 0x2 (gsm) read/write = 0x2 (gsm)/0x2 (gsm)

Hi, sorry to bother u all, i have a trouble when I call a did number forward to my asterisk server, the server told me: [Jul 28 19:00:57] WARNING[28080]: chan_sip.c:3806 sip_write: Asked to transmit frame type 256, while native formats is 0x4 (ulaw)(4) read/write = 0x4 (ulaw)(4)/0x4 (ulaw)(4) [Jul 28 19:00:57] WARNING[28080]: chan_sip.c:3806 sip_write: Asked to transmit frame type 4, while

Hi everyone, I have an issue which is kind of a catch 22 situation. I had outgoing calls to my new PSTN provider working perfectly. Then I started focussing on incoming calls. It seems that I can solve an error which gets my incoming calls working but that in turns means my outgoing calls don't work. - Strange. Anyhow I was getting an error: Process_sdp: No compatible codecs! And from

Hi there, I'll need some help with this: Trying to establish an IAX2 link between two servers works in one direction (SIP client with ulaw), but not in the other (SIP client with GSM). The client used for this is X-Lite behind NAT while both servers have a public IP (I playback an anouncement before trying to connect to the second *). Error on the originating * server:

We are getting the following on 1.8.3 and 1.8.4-rc2, HELP! Why is Asterisk unable to transcode to/from ulaw and g722? [2011-04-21 09:51:34] WARNING[22067]: chan_sip.c:6064 sip_write: Asked to transmit frame type ulaw, while native formats is 0x1000 (g722) read/write = 0x1000 (g722)/0x1000 (g722) [2011-04-21 09:51:34] WARNING[22067]: chan_sip.c:6064 sip_write: Asked to transmit frame type ulaw,

Hello! Client is using ulaw, however server sometimes fills the log with following: [2011-03-09 21:23:07] WARNING[27204] chan_sip.c: Asked to transmit frame type slin, while native formats is 0x8 (alaw) read/write = 0x4 (ulaw)/0x8 (alaw) [2011-03-09 21:23:07] WARNING[27204] chan_sip.c: Asked to transmit frame type slin, while native formats is 0x8 (alaw) read/write = 0x4 (ulaw)/0x8 (alaw)

Hi, When ever I put g729 in allow for trunk the other two codecs (ulaw and alaw) stop working and I get the frame type error for them, but g729 works fine. I've cleared general part of sip.conf of codec info to be on safe side. If ulaw and alaw are the only ones allowed they work fine. Asterisk shouldn't be doing any encoding or decoding, all codecs should be passing through. Any

Searching through wiki and google. http://www.2n.cz/products/gsm_gateways/voip/voiceblue.html but there are also other products on the market. --- Wondering if its possible to connect as follows: Extension -> Asterisk -> ZyxelAnalogTelephoneAdapter -> GSM gateway. The best way would be to make the ZyxelAnalog.. to be a channel. But I don't think that is doable.. or ? ---- So i

Hi, I am trying to achieve 3-way conferencing taking hint from wiki link http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO Here is the scenario: 1. user "ua1" calls user "ca1" 2. "ua1" then presses the feature code "*0" to redirect "ca1" to conference room 300 3. "ua1" then dials the user "33" 4. user

Hi All, I am stuck with an issue in the Openser+Asterisk Forking. In this solution we are using Openser as the Registrar. Hence it will store all the contact bindings along with the q values for a given user, say ua1. The current setup is such that the INVITEs are sent to Asterisk by Openser and Asterisk sends out the INVITE. Now if ua1 is registered with two different contacts having

Dear ALL: My scenario lists below: Assume: UA1 with sip id "1011" And dial number to PSTN is "0939749xxx" There is no modification rule at my CISCO. (It will not change any dialed number) UA1 ==> SER ==> UA2 (SIP to SIP) UA1 ==> SER ==> Asterisk ==> CISCO 5300 ==>

Hi, I have been experimenting with NAT and Asterisk a bit now. Though I have made progress along the way, I have come across the following problem. I'll really appreciate if anyone can provide me any help or pointers. Thanks! Successful Scenario: ------------------- All sorts of NAT calls are successful with full two-way media when both end points are locally subscribed users. Problem

Is there a way to get asterisk to send you a email when it looses or an extension doesn?t re-register Roger Workman Business Development Upperclassman/Universal Holdings LLC Voice: 304.324.3800 Fax: 304.324.3801 ICQ: 4447584 Website: http://www.upperclassman.net Billing Questions: billing at upperclassman.net Rental Questions: rentals at upperclassman.net Maintenance: help at

Is anyone getting '500 Internal Server' errors back from their Polycom phones when Asterisk sends a SIP NOTIFY message to them? I called Polycom tech support, who where utterly useless. Of course Polycom won't officially support it anyway, as they only support Asterisk Business Edition. We're using 1.2.9, but it's been ocurring for quite some time. We have about 35 phones and

Hi all, I did not get this error in Asterisk 1.2.5 release. I am testing on Asterisk SVN-trunk-r15187 to avail the PARKEDAT variable. - I park the call using ParkAndAnnounce - plays moh. - accept the call using ParkedCall The following errors are coming on the console and there is oneway audio - no audio after Music-On-Hold at caller's side. Please advice. I am testing using cisco 7902

Hi, We are using asterisk version 1.0.5. We have registered two UA's with asterisk. (Registration was successful) UA1 <-------> * <--------> UA2 Now, UA1 subscribes for UA2 to asterisk. asterisk sends NOTIFY to UA1 with UA2's state as open. But if UA2 gets un-registered then, asterisk is not sending NOTIFY to UA1. But when there is state change from UA2, asterisk is

We need to make sure that all queue applications run on the correct system that the user agents that own the queue application are registered to. So when a server fails and the user agents register with their secondary server (which will always be configured to be the same server for those related agents) the queue application is running on that server and routed to correctly by it's peers.

Hi, I am trying to integrate asterisk with openser for a simple call. I am facing some issues with Asterisk. Below is the explanation: I have a UA1 sending invite to UA2 through Openser and Asterisk with the below sequence. Sequence is UA1->OpenSER->Asterisk->Openser->UA2 When Asterisk gets the INVITE, the INVITE contains two Via headers, one of the UA1 and

This is not really a problem just something I noticed in my testing.. When two or more Asterisk servers are connected by IAX2 trunks it does not make use of any "shortest path" type system.. (maybe this is still planned somwhere down the line, but may come in handy to those who have multi asterisk installations) Here is the setup.. UA1--- Asterisk1----[IAX2 Trunk]---Asterisk2---UA2

Hi I have the following configuration | UA1 --|------ asterisk1 -----------------------+ UA2 --|------ asterisk2 -----------------------+ DB UA3 --|------ asterisk3 -----------------------+ UA4 --|------ asterisk4 -----------------------+ | All UA is located in the same area. A seperated PC is used as a centralized DB for storing a common dial plan, user account and register

Hello, I have problem to make SIP calls, i have asterisk + PC InterP4 + Digium TDM400P here is the content of the sip.conf: [SIP_PROVIDER] type=peer fromuser=testcomclient username=testcomclient secret=testr host=IP_SIP_PROVIDER ;dtmfmode=rfc2833 context=interne canreinvite=no ;allerid=Beer disallow=all allow=ulaw allow=gsm allow=g723.1 ; Asterisk only