Re: [asterisk-users] sip issue
:) On 31 October 2011 15:36, salaheddine elharit wrote: > thank you so much all works without issue now > > > > 2011/10/31 Christian Gansberger >> >> Hello, >> >> You have to disable RTP-Encryption on your Snom under Identity, RTP. >> It is set to on per default. >> >> >> On 31 October 2011 13:33, salaheddine elharit >> wrote: >> > hello list >> > >> > i have installed asterisk 1.8.7.1 and i have configured 2 account for >> > sip in >> > order to do internal call >> > >> > when i use x-lite and eyebeam1.5 i can call from 222 to 223 ,and alson >> > from >> > 223 to 222 >> > >> > but when i use my snom 320 i can call from my x-lite or eyebeam1.5 to >> > snom320 but the issue i can not call from my snom >> > >> > i have this issue just Asterisk 1.8 when i tested with asterisk 1.4 >> > theres >> > is no problem >> > >> > see the sip.conf and extenssions.conf below and also the cli when i try >> > to >> > call from my snom to x-lite >> > >> > thanks and regards >> > >> > CLI >> > == Using SIP RTP CoS mark 5 >> > [Oct 31 12:29:17] WARNING[16515]: chan_sip.c:8843 process_sdp: We are >> > requesting SRTP, but they responded without it! >> > salaheddine*CLI> >> > >> > sip.conf >> > >> > >> > [general] >> > context=agents >> > allowguest=yes >> > allowoverlap=no >> > allowtransfer=yes >> > allow=alaw >> > allow=ulaw >> > allow=gsm >> > allow=ilbc >> > [222] >> > type=friend >> > context=agents >> > host=dynamic >> > dtmfmode=auto >> > disallow=all >> > allow=alaw >> > allow=ulaw >> > qualify=yes >> > >> > >> > [223] >> > type=friend >> > context=agents >> > host=dynamic >> > dtmfmode=auto >> > disallow=all >> > allow=alaw >> > allow=ulaw >> > qualify=yes >> > >> > extenssions.conf >> > >> > >> > [agents] >> > >> > exten => 222,1,Dial(SIP/222) >> > exten => 222,n,Hangup() >> > exten => 223,1,Dial(SIP/223) >> > exten => 223,n,Hangup() >> > >> > -- >> > _ >> > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> > New to Asterisk? Join us for a live introductory webinar every Thurs: >> > http://www.asterisk.org/hello >> > >> > asterisk-users mailing list >> > To UNSUBSCRIBE or update options visit: >> > http://lists.digium.com/mailman/listinfo/asterisk-users >> > >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip issue
thank you so much all works without issue now 2011/10/31 Christian Gansberger > Hello, > > You have to disable RTP-Encryption on your Snom under Identity, RTP. > It is set to on per default. > > > On 31 October 2011 13:33, salaheddine elharit > wrote: > > hello list > > > > i have installed asterisk 1.8.7.1 and i have configured 2 account for > sip in > > order to do internal call > > > > when i use x-lite and eyebeam1.5 i can call from 222 to 223 ,and alson > from > > 223 to 222 > > > > but when i use my snom 320 i can call from my x-lite or eyebeam1.5 to > > snom320 but the issue i can not call from my snom > > > > i have this issue just Asterisk 1.8 when i tested with asterisk 1.4 > theres > > is no problem > > > > see the sip.conf and extenssions.conf below and also the cli when i try > to > > call from my snom to x-lite > > > > thanks and regards > > > > CLI > > == Using SIP RTP CoS mark 5 > > [Oct 31 12:29:17] WARNING[16515]: chan_sip.c:8843 process_sdp: We are > > requesting SRTP, but they responded without it! > > salaheddine*CLI> > > > > sip.conf > > > > > > [general] > > context=agents > > allowguest=yes > > allowoverlap=no > > allowtransfer=yes > > allow=alaw > > allow=ulaw > > allow=gsm > > allow=ilbc > > [222] > > type=friend > > context=agents > > host=dynamic > > dtmfmode=auto > > disallow=all > > allow=alaw > > allow=ulaw > > qualify=yes > > > > > > [223] > > type=friend > > context=agents > > host=dynamic > > dtmfmode=auto > > disallow=all > > allow=alaw > > allow=ulaw > > qualify=yes > > > > extenssions.conf > > > > > > [agents] > > > > exten => 222,1,Dial(SIP/222) > > exten => 222,n,Hangup() > > exten => 223,1,Dial(SIP/223) > > exten => 223,n,Hangup() > > > > -- > > _ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > New to Asterisk? Join us for a live introductory webinar every Thurs: > > http://www.asterisk.org/hello > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip issue
Hello, You have to disable RTP-Encryption on your Snom under Identity, RTP. It is set to on per default. On 31 October 2011 13:33, salaheddine elharit wrote: > hello list > > i have installed asterisk 1.8.7.1 and i have configured 2 account for sip in > order to do internal call > > when i use x-lite and eyebeam1.5 i can call from 222 to 223 ,and alson from > 223 to 222 > > but when i use my snom 320 i can call from my x-lite or eyebeam1.5 to > snom320 but the issue i can not call from my snom > > i have this issue just Asterisk 1.8 when i tested with asterisk 1.4 theres > is no problem > > see the sip.conf and extenssions.conf below and also the cli when i try to > call from my snom to x-lite > > thanks and regards > > CLI > == Using SIP RTP CoS mark 5 > [Oct 31 12:29:17] WARNING[16515]: chan_sip.c:8843 process_sdp: We are > requesting SRTP, but they responded without it! > salaheddine*CLI> > > sip.conf > > > [general] > context=agents > allowguest=yes > allowoverlap=no > allowtransfer=yes > allow=alaw > allow=ulaw > allow=gsm > allow=ilbc > [222] > type=friend > context=agents > host=dynamic > dtmfmode=auto > disallow=all > allow=alaw > allow=ulaw > qualify=yes > > > [223] > type=friend > context=agents > host=dynamic > dtmfmode=auto > disallow=all > allow=alaw > allow=ulaw > qualify=yes > > extenssions.conf > > > [agents] > > exten => 222,1,Dial(SIP/222) > exten => 222,n,Hangup() > exten => 223,1,Dial(SIP/223) > exten => 223,n,Hangup() > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Issue
You should set the ddwhome variable with the Set function or declare it on the global context. Try the Dial command with the dial string directly, before using the variable. Fro debugging purposes you should set debug and verbose at least to 10 and check the logs. Regards, Juan James A. Shigley wrote: > What do you mean I should use a global function. I'm kind both well versed > and a newb to * > > James Shigley > > > > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Juan E. > Rodríguez > Sent: Monday, December 28, 2009 12:44 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] SIP Issue > > Is ddwhome defined in global context?? If so, then you should use global > function. > > Paste asterisk log to check. > Saludos, > Juan E. Rodríguez > > > -Original Message- > From: "James A. Shigley" > Date: Mon, 28 Dec 2009 12:11:35 > To: Asterisk Users Mailing List - Non-Commercial > Discussion > Subject: [asterisk-users] SIP Issue > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Issue
What do you mean I should use a global function. I'm kind both well versed and a newb to * James Shigley -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Juan E. Rodríguez Sent: Monday, December 28, 2009 12:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP Issue Is ddwhome defined in global context?? If so, then you should use global function. Paste asterisk log to check. Saludos, Juan E. Rodríguez -Original Message- From: "James A. Shigley" Date: Mon, 28 Dec 2009 12:11:35 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] SIP Issue ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Issue
Is ddwhome defined in global context?? If so, then you should use global function. Paste asterisk log to check. Saludos, Juan E. Rodríguez -Original Message- From: "James A. Shigley" Date: Mon, 28 Dec 2009 12:11:35 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] SIP Issue ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Issue
On Mon, 2009-12-28 at 12:11 -0600, James A. Shigley wrote:
> Alright I have a SIP phone located off premises with a very annoying
> issue.
>
>
>
> Well I say a sip phone it is actually two phones hooked to a Cisco Spa
> 2102
>
> Link: http://www.cisco.com/en/US/products/ps10026/index.html
>
Looks pretty much like the PAP2 which I have running flawlessly with 1.6
in and outbound - so don't despair, you can solve this.
>
> Each phone being a different line/extension.
>
>
>
> Alright either line can ALWAYS make outbound calls no issue. The
> problem is on the Inbound side. I’m completely stumped as to why. I
> could make 10 back to back out bound calls and then call inbound and
> watch the call come in to * and try to be passed to the sip phone only
> to get “Error Message 14: Not a Working Number.” So it doesn’t seem to
> be a matter of the phones Sip Login “Timing out”
>
>
>
> And when I check sip peers it shows the correct IP address of the box
> so it doesn’t appear to be that it can’t find the Cisco box.
>
>
>
> Here is what I use for the inbound context, replacing the _X_ with the
> actual extension of course.
>
>
>
> [to_ddwhome]
>
> exten=> _X_,1,wait(1)
>
> exten=> _X_,n,Dial(${ddwhome},21)
>
> exten=> _X_,n,Goto(dial_inf,${EXTEN},1)
>
>
>
> ${ddwhome}=SIP/ddwhome
>
>
>
> Now the odd thing is when it gets the Error 14 message then the third
> step to dial_inf does not execute. Though when it rarely does connect
> with the sip phone if no one answers in 21 seconds than it will roll
> over to that step.
>
>
>
> Any ideas?
>
>
>
> James Shigley
>
Probably be useful to see sip.conf as well and know the version of
Asterisk you are running but in passing, you don't have any firewall
rules that could stop your asterisk talking TO the Cisco when something
comes in?
The one minor issue I had with mine is my router has some NAT issues
with signalling (It's a Draytek - they are known for it). In the end I
shifted the PAP2 up to 5061/5062 and the problem was gone. None of this
may be useful to you but I'll tell you this much. In my few weeks with
Asterisk I've had times where I've asked myself why certain things would
plain refuse to work and on every occasion it was *not* the fault of
Asterisk. 50% my config, 40% my network, 10% different docs for
different versions and missing info ;-)
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Re: [asterisk-users] sip issue with one way audio
Jason, What type of phones are you using? I originally started getting this error when I got the Cisco 7961Gs (prior to dumping them and going with all Polycoms). It turned out to be some setting in the XML provisioning boot file (although I can't remember which one). Once I went to a minimal config, the problem seemed to solve itself. Eventually I upgraded the SIP firmware and the problem disappeared regarless of the config file. Eric On Mon, 2007-08-06 at 23:38 -0600, Al lists wrote: > Nat? > > > On 8/6/07, Jason Walker <[EMAIL PROTECTED]> wrote: > I am getting this error > [Aug 6 15:28:26] WARNING[24452]: chan_sip.c:1920 retrans_pkt: > Maximum > retries exceeded on transmission [EMAIL PROTECTED] > for seqno > 102 (Critical Response) > [Aug 6 15:28:26] WARNING[24452]: chan_sip.c:1944 retrans_pkt: > Hanging > up call [EMAIL PROTECTED] - no reply to our critical > packet. > > any Ideas? > > Jason > > ___ > --Bandwidth and Colocation Provided by > http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Eric Lubow LinkExperts, Inc. Systems Administrator e: [EMAIL PROTECTED] w: www.linkexperts.com p: 212.542.5201 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip issue with one way audio
Nat? On 8/6/07, Jason Walker <[EMAIL PROTECTED]> wrote: > > I am getting this error > [Aug 6 15:28:26] WARNING[24452]: chan_sip.c:1920 retrans_pkt: Maximum > retries exceeded on transmission [EMAIL PROTECTED] for seqno > 102 (Critical Response) > [Aug 6 15:28:26] WARNING[24452]: chan_sip.c:1944 retrans_pkt: Hanging > up call [EMAIL PROTECTED] - no reply to our critical packet. > > any Ideas? > > Jason > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sip Issue
Michael,
Where in your extension definition to you dial a channel (SIP, Zap, or other)? You
are missing the dial entry.
-sb
-Original Message-
From: Lists [mailto:[EMAIL PROTECTED]
Sent: Saturday, November 29, 2003 10:53 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Sip Issue
Hi all I am having some issues with a gs 100 phone. It is on the same
network as my * server. There is no firewall.
In extentions.conf
exten => 5,1,Answer
exten => 5,2,MusicOnHold(default)
When I dial 5 from the sip phone
-- Executing Answer("SIP/mlh-2e75", "") in new stack
-- Executing MusicOnHold("SIP/mlh-2e75", "default") in new stack
-- Started music on hold, class 'default', on SIP/mlh-2e75
---about 7 secs...
-- Stopped music on hold on SIP/mlh-2e75
== Spawn extension (sip, 5, 2) exited non-zero on 'SIP/mlh-2e75'
In /var/log/asterisk/messages
Nov 29 23:01:46 WARNING[1142127920]: File chan_sip.c, Line 464
(retrans_pkt): Maximum retries exceeded on call
[EMAIL PROTECTED] for seqno 28503
(Response)
Any Ideas?
Michael
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