similar to: SIP->IAX->SIP silences

[I'm sorry to trouble the list with this, but this is the only way I know to contact the person concerned] This message is for Stephen Karrington - it appears that you have over-agressive 'spam' filters and we can no longer email you. Please rectify this if we are to have meaningful conversation! The original message was received from Linus Surguy

We have an Asterisk server that speaks IAX2 to Magrathea to get to the PSTN. Our local phones are a mix of Cisco 7940s and Grandstream BT100s all configured for SIP with silence-suppression disabled. Everything is configured to use a-law encoding. The version is: sip*CLI> show version Asterisk CVS-05/06/04-18:45:57 built by root@sip on a i686 running Linux Incoming callers are complaining of

I am looking to install a web interface for Asterisk to transfer calls and look who's on the phone. If anybody has a working web interface please let me know. I installed the www.asternic.com (operator) But when I bring up my web browser it says transferring data and does not bring a browser. -----Original Message----- From: asterisk-users-admin@lists.digium.com

HI all, Having a mental block today - can someone confirm which direction the echo cancelation applies to for the Zap PRI channels? ie. is it removing traces of the transmitted data to the PSTN from the received data, *or* is it removing traces of the data transmitted to Asterisk from the data received back from Asterisk? Got a configuration that is based on a call made from the PSTN to a SIP

I am considering using iax.cc (sixtel) and wondering if anyone had opinions, good or bad. Are there outages with any regularity? How responsive are tech support? How is packet loss? I am particularly interested in termination to the UK, but will accept any comments people have. Thanks -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US

Where can I investigate the origin of brief silences during calls from/to my SIP phone? Only rare pauses of 0.5-1.0 sec when I'm not able to hear anything. Thanks Mimmus

I am currently experiencing intermittent silences with my asterisk system. The symptoms are as follows: * Both for incoming and outgoing calls, I (and other users) occasionally experience a brief period of silence. * The silence lasts anywhere from 3 to 10 seconds. * It is not due to silence suppression, because the silences generally occur in the middle of sentences. * Silences occur at

Hi Community! If this issue was already topic, please excuse or delete my request... Topic 1 "no ringtone": I configured a SIP registration with my SIP provider (SIPCall). Everything works fine except the ring tone for the caller. The caller hears silence until the called party takes up the phone. I used the DIAL command with the r and R option but no luck... :( Has anybody the same

I have several of incoming numbers on IAX from voiptalk and magrathea but have a problem with IAX.conf. If I follow the example from voiptalk [VoIPTalk Incoming Number] type=friend username=VoIPTalk Incoming Number context=[XXXXXXXX] and make incoming entries in IAX.conf for the numbers like below with a different entry for each number pointing to a different context, incoming numbers always

Hi. [I asked a similar question a while back, but unfortunately wasn't around to reply to the responses, so sorry if you experience any deja vu.] I have a * server acting as an IVR system. The calls come in via IAX. After a period of about 40 seconds of silence (either waiting for the caller to dial an extension, or with the audio paused in controlplayback), the call hangs up. All I see in

Hi all, I'm getting some noise gate like effects on our sip lines & I think I need to disable silence supression, I'm searching docs & not finding where this can be set, does * have a setting to turn this off? basically what's happening is when we stop talking, the other end hears total silence, but when we talk, they can hear the background noise in the office, this sounds odd

DID's from Allstream (AT&T) are $2 Cdn/month but I think they have a rule that it has to terminate on their network somewhere... -----Original Message----- From: Linus Surguy [mailto:linus@magrathea-telecom.co.uk] Sent: Monday, June 14, 2004 6:53 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Canadian DID Can anyone point me in the direction of a wholesaler of

Any recommendations for a VoIP provider in the UK? I have a few guys in a field office in the UK with SIP phones and a VPN tunnel back to a working Asterisk setup in the US. The Asterisk setup has an IAX trunk with TelaSIP/VoipXpress with local DiD's for US offices, so they can call vendors, customers etc in the US at local rates. I'd like to get the same thing for the UK, so that UK

Hi All, Is there anyone providing UK geographic numbers that can be terminated on Asterisk using IAX ? It must be a geographic number (eg. Start 01 or 02, not 08xx). I've tried the sipcall.co.uk service and it looks very good when using X-Lite but it will not work with Asterisk. Switching to IAX should also resolve issues around NAT - hurray! -Nathan

Hi, Currently using voiptalk.org and the quality is getting really bad. I would like a second provider preferably in UK, anyone got any suggestions? Ta. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040421/3d91c7f6/attachment.htm

Been playing with G.722 in Asterisk 1.4.24.1 - using the back-ported patches from http://carlton.oriley.net/drupal/node/12 Works just fine as far as I can tell - Grandstream phones anyway - playing the G722 sound files, and calls between them. Transcoding seems fine too - calling non G722 devices, it seems to "just work" However Phone A (g722) calls phone B (gsm). Works fine.

Hi all, I am experiencing a problem with the quality of the voice communication between an IAX based softphone (WinIAX) and an outside line through a FXO port or even with a regular analog phone connected to a FXS port. The party using the IAX softphone hears his own echo a plit of a second after speaking. The party on the analog end does not experience any echo. I tried to modify the KFLAG

On 03/11/03 00:25, Mark Spencer wrote: > As a side note, I strongly would like to see someone implement a > client using libiax2 which implements IAX2 instead of the (now > obsolescent) IAX version 1. I'm implementing a Java-based IVR server (and yes, I know Asterisk does IVR, and no, it's not flexible enough to do what I want and no, it doesn't integrate well with the Java

I seem to be having problems with my 2 asterisk systems.. This works.. Call originating from the PSTN and then routed to the UA.. PSTN-->[pbx1]--IAX-->[pbx2]--SIP-->UA This doesn't work.. Call originating from UA to PSTN.. UA--SIP-->[pbx2]--IAX-->[pbx1]-->PSTN When the call is placed FROM the UA to the PSTN the call goes through and the PSTN phone rings but when it is

Ok.. I have done some more digging and the problem seems to be caused by chan_capi not detecting that the call has been answered.. I downgraded chan_capi from 0.2.3b to 0.2.2 and the system is working fine.. Kapjod, do you know of this problem? Later.. > I seem to be having problems with my 2 asterisk systems.. > > This works.. Call originating from the PSTN and then routed to the