similar to: a=fmtp:18 annexb=no

Hi all, I am using asterisk1.2.14,realtime and I find there is a strange case in the receiver's display. I have a dial plan to route a call to the destination. I haven't set the callerid(num) for the caller. In the receive ends, it's display shows "asterisk" when I make a call to the receiver. I wonder why "asterisk" shows in the display as I haven't set

I've been running the 8/1/2004 Head release up until a little over a week ago. I was forced to due to a card failure to upgrade to 1.2.16 without any advance preparation or testing (most of my connections are via satellite to all corners of the globe with high latency). Up until the upgrade I was running with very few issues. Since the upgrade I have been experiencing strange issues

? HI ? I have asterisk 1.6.13 running. My ATA are connected via vpn (openvpn) ? Out going calls from asterisk to the ata works fine Incoming calls from the ata to asterisk cuts off with the error msg ? Maximum retries exceeded on transmission 70854efe-4157e3a8 at 10.168.7.103 for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt. [Dec? 6 13:52:43] WARNING[3921]: chan_sip.c:3858

Skipped content of type multipart/alternative-------------- next part -------------- Nov 21 14:17:47 VERBOSE[32580] logger.c: <-- SIP read from 172.25.103.222:5060: INVITE sip:1715;phone-context=exp_net.ascom@ascom.be:5060;maddr=172.25.96.48;transport=udp;user=phone;x-nt-redirect=redirect-server SIP/2.0 From:

Specs: Asterisk 1.4.22 running behind a SonicWall (transparent mode) with a public IP address. We have our phone system setup as 172.16.2.x that connect through the SonicWall to Asterisk. Incoming calls work flawlessly and we no longer get one-way audio. We are only using SIP (3 trunks now, instead of 2) and having all 3 in use is not an issue. Problem: Make a call on a Polycom 320 IP phone to

I am debugging an intermittent issue of missing audio on calls that come from a SIP provider into our asterisk-11.10 installation. Sometimes, incoming calls from this provider work correctly, with audio streaming in both directions. Other times, with the same setup, the calling party is unable to hear the IVR recording from the asterisk installation, although in fact the streaming is supposed to

I have a asterisk server running on site listening on a public ip. Tonight I attempted to connect a Cisco 7960 phone from my home location via sip but failed. My home network is simple, Cox cable connection hooked to a linksys wrt router. The firewall on the linksys router is disabled and I even setup dmz to the phones ip as a last resort. I removed the linksys router and plugged the phone

Hello Everyone, Today we observed asterisk sending two invites for the initial call before the call was established (ie, not re-invites). There were no changes made to the configuration for a very long time, and was kind of confused when seeing this action. Can someone please suggest where to look to remove this behaviour? U 2014/08/12 07:34:20.405029 192.168.2.10:5060 -> 192.168.2.20:5080

On Tue, Apr 9, 2013 at 1:22 PM, Bogdan-Andrei Iancu <bogdan at opensips.org>wrote: > ** > Hi Nick, > > The BYE is not properly formed and rejected by script - in the 200 OK of > the INVITE, you can see that your opensips is doing Record-Routing, but the > BYE does not contain the corresponding Route hdr, so SIP routing is > impossible. > > Regards, > >

another issues with cisco 7975 I have phone registered on asterisk have 2 different issues on different versions of firmware, on 9-4-2-1S I have not working 3way conference, when I trying to connect second call, phone says ?unable to set up conference? and sending some cisco xml data to asterisk which cannot be handled, thats the problem, I know on firmware 8-5-4 3way conference works just

Asterisk Project Security Advisory - AST-2018-003 Product Asterisk Summary Crash with an invalid SDP fmtp attribute Nature of Advisory Remote crash Susceptibility Remote Authenticated Sessions Severity Minor

Hello, I'm having this issue on my pbx, it appears that asterisk is refusing the codecs that my providers is proposing. My trunk configuration is: --- username=5x5x7x9x0x3 type=friend secret=CRcxn7sqwm qualify=yes port=5060 insecure=port,invite host=sip.txtxlxoxp.it fromuser=5x5x7x9x0x3 fromdomain=sip.txtxlxoxp.it disallow=all context=from-trunk allow=alaw --- A typical invite from my

I am not sure what you mean by the ful SIP signaling? Here is the trace for the sip.conf which works successfully. Below that, I will include the trace for the pjsip.conf which it seems Vitelity isn't accepting the ACK in response to the OK ---- SIP --- <--- Transmitting SIP request (1004 bytes) to UDP:64.2.142.93:5060 ---> INVITE sip:8005555555 at 64.2.142.93 SIP/2.0 Via: SIP/2.0/UDP

I've got a fresh (1 day old) svn trunk release SVN-trunk-r225360 of Asterisk and I'm trying to get it to accept a SIREN14 call from Polycom's softphone. Having trouble with SDP negotiation, I want to only allow SIREN14 and nothing else. I also want to record and playback files, any tips on what the Record function parameters should be? In sip.conf I have: disallow=all

On about 25% of inbound calls to a ring group, picking up any one extension as it rings results in dead air. Some details regarding my VoIP network to make the following logs more readable: 192.168.7.130 resolves to the trixbox host. 192.168.7.135 resolves to endpoint 812. 192.168.7.137 resolves to endpoint 811. 192.168.7.138 resolves to endpoint 810. 192.168.7.139 resolves to endpoint 813.

Hi, I have set up an asterisk with TLS and SRTP support. The SRTP is working with Phonerlite softphone. I have problem with the SRTP, when I make calls on Audiocodes gateway . I got the folloowing messages on asterisk: [Oct 2 10:59:48] NOTICE[24868]: sdp_crypto.c:232 sdp_crypto_process: Crypto life time unsupported: crypto:1 AES_CM_128_HMAC_SHA1_80

Hi, I was able to use Unsniff to validate that the incoming 20 byte payloads of audio from the downstream IAX2 trunk was definitely G.729a whilst Asterisk 16.13.0 transcodes to G.711a unnecessarily. Media is confirmed as having been negotiated as g729 on all four streams. Nuance with this call is that it's from a WebRTC client which doesn't transmit any audio, could this be influencing

Hi. I'm stuck into an odd situation. Here's what happens: 4 Thomson ST2030S 2 Cisco 7912 3 Cisco 7940 2 AAstra 480i Asterisk 1.2.17 Diva 4BRI + chan_capi I've just upgraded yesterday from Asterisk 1.2.13 to 1.2.17. Until yesterday, everything was just fine with 1.2.13. Immediately after the upgrade, *all* the 7940 are no more able to make calls, just receive them, while 7912

Hello. Has anybody knows how run two asterisk process in one hardware? (each one with its own configuration?) Thanks Juan Salas.

I'm communicating a softphone (SJPhone) to a Grandstream phone GXP-2000. The codec order on each one is the next: SJPhone: GSM - iLBC - PCMA - PCMU GXP2000: G729 - GSM - PCMA - PCMU (I have a G729 license, so there's no problem with transcoding G729) In my sip.conf, I've defined the following codec order: disallow=all allow=g729 allow=gsm allow=g726 allow=alaw allow=ulaw And my