similar to: How to integrate thirdparty RTP with Asterisk

Hello My asterisk is stoping. i am using asterisk with ser on same mechine here is the asterisk trace ------------------------------------------------ -- Setting call duration limit to 3000 seconds. Sep 2 15:58:12 WARNING[10334]: rtp.c:852 ast_rtp_new_with_bindaddr: Unable to allocate socket: Too many open files Sep 2 15:58:12 WARNING[10334]: chan_sip.c:2313 sip_alloc: Unable to create RTP

Hi, I have a Intel Xeon Dual Core server, with 3 GB RAM, running Centos 5.0, Asterisk 1.4.4 and mysql 5.0. It is a kinda high-traffic box, with about 60 concurrent calls. The profile of calls on this box are: Incoming: via a Sangoma A101 via SIP from anothjer SIP server Outgoing all calls that come in are sent out via SIP to yet another SIP server. This morning I has this error: (edited)

This was a bug 1.4.4 It has now been fixed in Asterisk 1.4.5 Stuart Bennett wrote: > Hi Yusuf > > A friend of mine had the same problem with a high volume site.. The problem > lies with a limitation in Linux. Linux will only allow a certain amount of > open files at a time. You will need to add the following line before running > asterisk. > > ulimit -n 32768 > >

My asterisk process produced the following errors this morning: Dec 8 10:44:07 WARNING[50315282]: rtp.c:829 ast_rtp_new_with_bindaddr: Unable to allocate socket: Too many open files Dec 8 10:44:07 WARNING[50315282]: chan_sip.c:2352 sip_alloc: Unable to create RTP session: Too many open files Dec 8 10:44:07 WARNING[50315282]: chan_sip.c:8024 sip_request: Unable to build sip pvt data for

Dear all, we have encounter problem when starting asterisk in the foreground, "asterisk -vvvvgc" with more 100 SIP calls concurrently. we have set ulimit to the highest value. still has this problem. Is this the problem keeping asterisk in the foreground or this is a bug in SVN 1.2 16771? Apr 5 08:48:36 WARNING[14887]: channel.c:562 ast_channel_alloc: Channel allocation

Hi, I am trying to build asterisk with xtensa compiler on embedded platform. I am trying to integrate my driver code to asterisk. For this tying to call driver code IOCTLs from chan_dahdi instead of dahdi IOCTLs. While compiling asterisk with xtensa, Observed chan_dahdi is not compiling [chan_dahdi.so is not creating].But with default settings [/usrc/src] it was creating. Please let me

Hello all, It used to work on calling my GTalk ID from another GTalk user. But now that I tried calling it again, the caller hears only a ringtone and disconnected after a few rings. The messages on my Asterisk-1.4.21.2 are the following: [Jan 29 10:37:51] ERROR[1303]: rtp.c:1945 ast_rtp_new_with_bindaddr: Unexpected bind error: Cannot assign requested address [Jan 29 10:37:51] WARNING[1303]:

Hi Jack, > You're comments are welcome. Here they are... 3.1 RTP Header Payload Type (PT): I don't see an alternative to using a value of the dynamic range (96-127). IIRC, other ranger are reserved for fixed values assigned by IETF. "A dynamic payload type MUST be used - i.e., one in the range [96,127]." 3.2 Payload Header If you refer to RFC2119, please keep the capital

Hello, we're in process of testing an interconnection with a trans-european carrier. But the carrier wants the SIP signalling to server 1 and the RTP stream to server 2. How do I configure asterisk to work with that type of installation. It seems they are using NexTone as SIP Signaling and RTP servers. Can someone help me??? Regards, Marc -- CTO Marc Storck

Hello List, I need your advise please. I am trying to establish a call from Asterisk 1.8.15-cert5 to one remote SIP UA (not Asterisk), both are behind NAT. That remote peer is configured with nat=yes in my sip.conf but yet RTP packets are being sent to its internal IP address which is declared in the Connection Information (c) in the SDP, obviously reaching nowhere. I need RTP to be sent to the

Hello List, I need your advise please. I am trying to establish a call from Asterisk 1.8.15-cert5 to one remote SIP UA (not Asterisk), both are behind NAT. That remote peer is configured with nat=yes in my sip.conf but yet RTP packets are being sent to its internal IP address which is declared in the Connection Information (c) in the SDP, obviously reaching nowhere. I need RTP to be sent to the

Hi I have a weird issue with a new 1.6.2.17.2 box. At random intervals it just stops responding to RTP and signalling (both SIP and IAX observed). All calls in progress lose audio both ways although the console shows the call legs still in progress. No signalling can be sent or is received. It is as though the server drops of the net for those protocols. I can still navigate the console. Killing

Good day. I'm trying to configure termination with The Asterisk thru Cisco AS5300 Gateway from the SIP softphone (X-Ten X-Lite) to POTS network. I think, I had recognise kind of problem: call is ringing in the POTS phone (so I guess SIP signalling is working ok?), but there is no voice in either sides. On the Asterisk PC I can see incoming RTP streams with tcpdump and tethereal, but I

I'm confronted with an issue that I am sure many others are too with Asterisk and scalability. I'd like to be able to build a cluster of Asterisk boxes to handle a large volume of simultaneous calls but have the feeling that the hardware requirements to handle large volumes of RTP streams would be too vast. So with that assumption I imagine a platform that would not get involved with the

Hi, Thanks to those who hinted on the SIP/H323/Skinny capabilities of asterisk ... I am starting to like this app! :D Now, I successfully managed to bridge SIP to H323 (i don't have skinny phones here). Just a question: Is it possible to have Asterisk "just" as a signalling proxy? i have a flat test network, and i would like the RTP streams to be sent directly end to end (sip phone

I'm currently working with Asterisk 11.8.1 trying to get Multicast RTP working (it's not) with some Polycom phones, and I'm really trying to determine if Asterisk or the phones are the issue. I THINK it's Asterisk... In extensions.conf I have a simple: "Page(MulticastRTP/basic/x.x.x.x:xxxx) line, and when I dial that extension I get: -- Called

FYI, the Opus RTP payload format is now RFC7587: https://tools.ietf.org/html/rfc7587 Cheers, Jean-Marc -------- Forwarded Message -------- Subject: [payload] RFC 7587 on RTP Payload Format for the Opus Speech and Audio Codec Date: Tue, 30 Jun 2015 16:33:17 -0700 (PDT) From: rfc-editor at rfc-editor.org To: ietf-announce at ietf.org, rfc-dist at rfc-editor.org CC: drafts-update-ref at

Hi All, I would like to separate the media traffic from the signalling. Can Asterisk send and receive media (rtp) traffic from a secondary network interface? Thanks, Harel ________________________________ This electronic message and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you are not the named

Hi Jean-Marc, Just some initial feedback, I've just tried to build svn head using the Visual Studio 2005 compiler, and had the following issues: 1. Missing definition of M_PI if it's undefined (and it is on this platform) 2. In speex_resampler_process_int, the compliler can't determine the value of *in_len and *out_len at compile time and thus determine the size of the x and y

Greetings! I have an Asterisk 1.4 box and due to hardware limitations I cannot upgrade atm. So, as long as I understood from different posts, SIP-TLS is not available for 1.4 Then I set up VPN and route all inter-Asterisk traffic into VPN. But for some reason, with all the RTP inside the VPN I start getting packet losses up to 30%. Maybe CPU is too weak, that is yet to be discovered. What