similar to: Send SIP 100 Trying instead of 183 Session Progress

Displaying 20 results from an estimated 600 matches similar to: "Send SIP 100 Trying instead of 183 Session Progress"

2007 Jul 31
1
Turn off SIP 183 Session Progress in Asterisk 1.4.8
[Resent due to non-descriptive subject line.] Hi folks When connecting two SIP users, is there any way to stop Asterisk from sending SIP 183 Session Progress messages, either globally or per-peer? Scenario as follows: Call from UA1 to Asterisk (UA2) to UA3. UA3 sends RTP before SIP OK to Asterisk (UA2). Asterisk (UA2) detects early audio from UA3 and sends 183 Session Progress with SDP to
2011 Apr 21
1
Transcode ulaw/g722 problem
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,
2010 Nov 12
1
Call failed becaus of SIP tanslate
Hi Guys, I have a the following issue when I ma trying to place a call through my voip provider, I am using an asterisk 1.2.21.1, do you have an idea what could fix this issue (as you can see when the other party answered, the call get dropped because of probably sip incompatibility) Nov 12 14:31:30 WARNING[21432]: chan_sip.c:2587 sip_write: Asked to transmit frame type 256, while native formats
2011 Nov 11
3
1.8.7.0 crashing : Can't send 10 type frames with SIP write
With asterisk 1.8.7.0 has been running ok for months. Now, this morning, it's crashing. I can restart it, but it crashes after 10+ minutes. It dies like this -- Executing [s at macro-stdexten:2] Dial("SIP/teliax-00000019", "SIP/176,18,rtT") in new stack == Using UDPTL TOS bits 184 == Using UDPTL CoS mark 5 == Using SIP RTP TOS bits 184 == Using SIP RTP
2015 Jul 15
2
Problem "no voice"
Hi list! I have 4 numbers on my Asterisk 1.8. 3 work perfectly, the 4.th not. I'm not sure, when it finish to work, since a month ago it runs without any problem... Well, if I will be called on this number I can't hear anything and in Asterisk I see these: [Jul 15 18:59:55] WARNING[8752]: channel.c:5060 ast_write: Codec mismatch on channel SIP/00493514977290-000001d1 setting write format
2015 Feb 23
2
Question about Warning message
Starting with Asterisk 13.1 we are seeing this WARNING messages a lot in our logs and console: WARNING[25164][C-0004865e]: chan_sip.c:7364 sip_write: Can't send 10 type frames with SIP write) We found that line in function "sip_write" inside "chan_sip.c". In our previous version (11.2.1) we did not see those messages being printed (same verbosity level). We compared
2005 Jun 08
2
format g729 and Voxee.com
Hi, I have just signed up with Voxee.com and have attached my Asterisk server to dial them via IAX2. Below is the start of the log which dials the number and promply hangs up when the call is answered, with the logs saying that the channel is not compatiable. I have traced this down to the g.729 codec which I don't have installed. Any ideas on how to force that the codec not be used?
2006 Mar 22
5
Double Call Progress tones
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 This is slowly driving me nuts! I have several Cisco 7960s with SIP 8.2/7.5 fw connecting to Asterisk 1.2.5 driving a TE110P on a BT EuroISDN PRI line. On all outgoing calls I get a double ring tone (UK style + US style). I also have a DECT phone on a Sipura SPA-3000 configured with UK tones. This gives me a double ring of UK + UK, so this
2016 May 03
2
Migrating asterisk 11 to 13: some callers get no ringback tone any more
Whoops, email client auto-filled dev previously. Let's try this again. Michael Maier wrote: <snip> > Ok - but this doesn't seem to answer my main question: > > Why must > > progressinband=never > > be applied especially if asterisk uses it by default? The big difference > between w/ and w/o it is: The default in 13 is "no" which still
2010 Jun 11
3
no ring back 180 with SDP
I have a box (Genband) expecting the following: 100 trying 180 ringing with SDP Or 100 trying 183 with SDP And asterisk is sending: 100 trying 180 ringing 183 with SDP Any way to modify asterisk to send what he is expecting? Thanks, Dave -------------- next part -------------- An HTML attachment was scrubbed... URL:
2016 May 03
3
Migrating asterisk 11 to 13: some callers get no ringback tone any more
Hello! I migrated asterisk 11 to 13 as user of FreePBX 12.0.76.2. As customer of German Telekom, I have three numbers and therefore three trunks - each number is bound to one trunk. After migration, some callers complained about missing ringback tone: they didn't hear any ring tone and where surprised that they suddenly got me anyway :-). The connection afterwards was as expected. Deeper
2005 Jul 13
1
Suddenly a problem with outgoing calls made from Cisco phones...
Hi all! Quite a mystery. The following happened when I was on holiday, and no one else has changed any configs of either Asterisk or the Cisco's in the building... The situation: Incoming works fine on all phones. Outgoing only works from non-Cisco phones. When calling from a Cisco phone to an external phone, all the Cisco-user hears is a ticking crackle and after about a minute the phone
2011 Apr 07
3
No ringback even though progressinband=yes is set
Any ideas on why callers who call into my customer's SIP trunk are not hearing a ringback tone? I had this on one other asterisk system, and wound up needing to set progressinband=yes in the SIP trunk config. I have set this on the current system & restarted asterisk, but to no avail. I am using: AsteriskNOW distro Asterisk build is 1.6 from AsteriskNOW repository:
2005 Mar 02
1
Getting Polycom IP500 to talk to Asterisk - um... Newbie question :)
> Hmmm... I have this aweful feeling that I'm choosing the > exact wrong time to ask a "newbie question" :) Oh well, here > it goes. > > The quick question is : "How do I dial an extension?" > (answer is probably - "you don't" in which case:) "How do I > dial my asterisk box?" - I have no outside line, I just want >
2006 Dec 11
1
Asterisk Sends 180-RINGING to UA even with progressinband=yes
I have progressinband=yes in sip.conf, but Asterisk sends a 180-Ringing to my polycom phones and then it also sends 183-Session Progress. That doesn't seem to make sense. Shouldn't Asterisk NOT send 180-Ringing if progressinband=yes ? Doug.
2019 Jun 14
2
Early Media Issue
Hi all I've got an issue where when I call a number that just plays early media back to me. Instead of hearing the full sequence of tones I hear a short ringing then part of the sequence. What seems odd is that I can see the telephone-event/8000 being passed up the chain but when it gets to Asterisk, it is never sent back to the phone. Instead I just see the usual RTP flows. I've been
2009 Jun 13
2
Polycom registration errors
I'm evaluating using Polycom phones for our call center and I've set up my first phone (a SoundPoint 560) to give it a try. The phone is working and can successfully place and receive calls. But every minute, there's an error in the log file: chan_sip.c: Registration from '<sip:6193644850 at jtsd05>' failed for '192.168.200.99' - Username/auth name
2004 Dec 28
1
Asterisk / 183 message
Hello, My company is doing some * testing with our Class 5 softswitch and had some questions regarding ringback being provided to our PSTN users (off --> on net calling) Currently with MGCP subscribers, we know the PSTN ringing is provided by a digital PBX for example, However, it looks like with SIP, our softswitch is relying on MGCP signaling on our PSTN gateways to provide ringback
2006 Feb 23
3
Polycom IP601 Question
Hey everyone, I haven't seen an issue quite like mine, so I am hoping anyone who used the Polycom 601's may have an idea. We are going to be switching our office over to Asterisk. All the phones are going to be 601's, I am going to set up a boot server, but for now I am just going to test everything on one phone. My question is I have the phone registered in Asterisk (phone icon
2010 Jul 20
3
Problem with SIP
Good afternoon list. I'm experiencing a problem with my SIP channel's. When I have an external connection for one of my SIP carrier's, I can listen to the client and the client listens to me normally. The problem is when I will transfer this connection, the call is mute for the extension I have transfered. Only the client hears normally. In the console of Asterisk generates the